CS6304 Notes

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UNIT I FUNDAMENTALS OF ANALOG COMMUNICATION
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UNIT II DIGITAL COMMUNICATION

INTRODUCTION:
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FREQUENCY SHIFT KEYING:
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UNIT III DIGITAL TRANSMISSION

INTRODUCTION:
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Delta Modulation (DM) :

Delta Modulation (DM) :
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Let m[ n ] = m(nT s ) , n = 0,±1,±2, 

where Ts is the sampling period and m( nT s ) is a sample of m(t ).
The error signal is
e[ n ] = m[ n ] − mq [ n −1] (3.52)
eq [ n ] = ∆sgn( e[ n ] ) (3.53)
mq [ n ] = mq [ n −1] +eq [ n ] (3.54)
where mq [ n ] is the quantizer output , eq [ n ] is
the quantized version of e[ n ] , and ∆is the step size
The modulator consists of a comparator, a quantizer, and an accumulator. The output of

the accumulator is
Differential Pulse-Code Modulation (DPCM):

Usually PCM has the sampling rate higher than the Nyquist rate .The encode signal
contains redundant information. DPCM can efficiently remove this redundancy.
Figure 3.28 DPCM system. (a) Transmitter. (b) Receiver.
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e[n ] = m[n] −m ˆ [n ] (3.74)
mˆ [n] is a prediction
value.
The quantizer output is
eq [n] = e[n] + q[n] (3.75)
where q[n] is quantizati on error.
The prediction filter input is
mq [ n] = m
ˆ [n ] +e[n] +q[n] (3.77)
m[n ]
⇒mq [n] = m[ n] + q[ n] (3.78)
Processing Gain:
The (SNR) o of the DPCM system is

σ M2
(SNR) = (3.79)
σQ2
o
where σ M2 and σQ2 are variances of m[ n ] ( E[ m[ n]] = 0) and q[ n ]

σ M2 σ E2
(SNR) =( )( )
σ E2 σQ2
o
= G p (SNR ) Q (3.80)
where σ E2 is the variance of the prediction s error
and the signal - to - quantizati on noise ratio is
σ E2
(SNR ) Q = (3.81)
σQ2
σ M2
Processing Gain, G p = (3.82)
σ E2
Design a prediction filter to maximize G p (minimize σ E2 )
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NEAR EAST UNIVERSITY

FACULTY OF ENGINEERING
DEPARTMENT OF COMPUTER ENGINEERING
COM 318
DATA COMMUNICATIONS
LECTURE NOTES
Prepared by
Dr. Tayseer Alshanableh
Nicosia-2007
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COM 318-Ch.I-IV 07/08 Fall
TABLE OF CONTENTS
CH. 1: INTRODUCTION TO DATA COMMUNICATIONS

CH. 2: DATA TRANSMISSION & SIGNALS
CH. 3: TRANSMISSION MEDIA
CH. 4: ENCODING, MODULATING & TRANSMISSION CODES
CH. 5: TRANSMISSION OF DIGITAL DATA: INTERFACES & MODEMS
CH. 6: MULTIPLEXING
CH. 7: ERROR DETECTION AND CORRECTION
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CHAPTER 1
INTRODUCTION TO DATA COMMUNICATIONS
- COMPUTER NETWORK
Interconnected collection of autonomous computers that are able to exchange information.
 No master/slave relationship between computers in the network.
- DATA COMMUNICATIONS
Transmission of signals in a reliable and efficient matter.
- COMMUNICATION MODEL (SYSTEM)
The purpose of a communications system is to exchange data between two entities.
 Source: entity that generates data; eg. a person who speaks into the phone, or a
computer sending data to the modem.
 Transmitter: a device to transform/encode the signal generated by the source.
- the transformed signal is actually sent over the transmission system.
eg. a modem transforms digital data to analog signal that can be handled by the
telephone network.
 Transmission System (Channel): medium that allows the transfer of a signal from
one point to another.
eg. a telephone network for a computer/modem.
 Receiver: a device to decode the received signal for handling by destination device.
eg. A modem converts the received analog data back to digital for the use by the
computer.
 Destination: entity that finally uses the data.
eg. Computer on other end of a receiving modem.
Data Communications
Data communications is the transfer of information that is in digital form, before it enters the
communication system.
- Basic Elements of a Communication System
Signal s(t) Channel r(t)
Transmitter Receiver
n(t)
Information
source & input
Noise Output
transducer transducer
Basic Elements of a Communication System

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 Information: generated by the source may be in the form of voice, a picture or a plain
text. An essential feature of any source that generates information is that its output is
described in probabilistic terms; that is, the output is not deterministic.
A transducer is usually required to convert the output of a source in an electrical signal
that is suitable for transmission.
 Transmitter: a transmitter converts the electrical signal into a form that is suitable for
transmission through the physical channel or transmission medium. In general, a
transmitter performs the matching of the message signal to the channel by a process
called modulation.
The choice of the type of modulation is based on several factors, such as:
- the amount of bandwidth allocated,
- the type of noise and interference that the signal encounters in transmission over the
channel,
- and the electronic devices that are available for signal amplification prior to
transmission.
 Channel: the communication channel is the physical medium that connects the
transmitter to the receiver. The physical channel may be a pair of wires that carry the
electrical signals, or an optical fibre that carries the information on a modulated light
beam or free space at which the information-bearing signal are electromagnetic waves.
 Receiver: the function of a receiver is to recover the message signal contained in the
received signal. The main operations performed by a receiver are demodulation,
filtering and decoding.
Analog and Digital Data Transmission
- Data are entries that convey information.
- Signals are electrical encoding (representation) of data.
- Signalling is the act of propagation of signals through a suitable medium.
The terms analog and digital correspond to continuous and discrete, respectively. These
two terms are frequently used in data communications.
Analog data takes on continuous values on some interval. The most familiar example of
analog data is audio signal. Frequency components of speech may be found between
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20 Hz and 20 kHz. The basic speech energy is concentrated between 300-3400 Hz. The
frequencies up to 4000 Hz add very little to the intelligibility of human ear.
Another common example of analog data is video. The outputs of many sensors, such as
temperature and pressure sensors, are also examples of analog data.
Digital data takes on discrete values; eg. a computer’s output.

- Analog transmission is a means of transmitting analog signals regardless of their
content. The data may be analog or digital.
- Digital transmission is the transfer of information through a medium in digital form. A
digital signal can be transmitted only for a limited distance.
- Data communications is the transfer of information that is in digital form, before it
enters the communication system.
 Two methods of sending data from computer A to computer B. both cases are
examples of data communications, because the original data is digital in nature.
Digital
Source Modem Modem
A Analog Transmission
A B
Digital Transmission
 Two ways of transmitting analog information. In either cases it is not data

communications, because the original information is not digital.
Digital Transmission
ADC DAC
Analog Destination
Source
Analog Transmission
ADC: Analog-Digital-Converter
DAC: Digital-Analog-Converter
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Digital Communication System
Up to this point, we have described an electrical communication system in rather broad terms
based on the implicit assumption that the message signal is a continuous time-varying
waveform. We refer to such continuous-time signal waveforms as analog signals and to the
corresponding information sources that produce such signals as analog sources. Analog
signals can be transmitted directly via carrier modulation over the communication channel
and demodulated accordingly at the receiver. We call such a communication system an analog
communication system.
Alternatively, an analog source output may be converted into a digital form and the message
can be transmitted via digital modulation and demodulated as a digital signal at the receiver.
There are some potential advantages to transmitting an analog signal by means of digital
modulation. The most important reason is that signal fidelity is better controlled through
digital transmission than analog transmission. In particular, digital transmission allows us to
regenerate the digital signal in long-distance transmission, thus eliminating effects of noise at
each regeneration point. In contrast, the noise added in analog transmission is amplified
analog with the signal when amplifiers are used periodically to boost the signal level in long-
distance transmission. Another reason for choosing digital transmission over analog is that the
analog message signal may be highly redundant. With digital processing, redundancy may be
removed prior to modulation, thus conserving channel bandwidth. Yet a third reason may be
that digital communication systems are often cheaper to implement.
In some applications, the information to be transmitted is inherently digital, e.g., in the form
of English text, computer data, etc. In such cases, the information source that generates the
data is called a discrete (digital) source.
In a digital communication system, the functional operations performed at the transmitter and
receiver must be expanded to include message signal discrimination at the transmitter and
message signal synthesis or interpolation at the receiver. Additional functions include
redundancy removal, and channel coding and decoding.
Figure 1.2 illustrates the functional diagram and the basic elements of a digital
communication system. The source output may be either an analog signal, such as audio or
video signal, or a digital signal, such as the output of a Teletype machine, which is discrete in
time and has a finite number of output characters. In a digital communication system, the
messages produced by the source are usually converted into a sequence of binary digits.
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Information
Source Channel Digital
source & input
encoder encoder modulator
transducer
Data Communication System Channel
Output Output Source Channel Digital

transducer decoder decoder demodulator
Signal
Figure 1.2. Basic elements of a digital communication system

Ideally, we would like to represent the source output (message) by as few binary digits as
possible. In other words, we seek an efficient representation of the source output that results
in little or no redundancy. The process of efficiently converting the output of either an analog
or a digital source into a sequence of binary digits is called source encoder or data
compression.
The sequence of binary digits from the source encoder, which we call the information
sequence, is passed to the channel encoder. The purpose of the channel encoder is to introduce
in a controlled manner some redundancy in the binary information sequence, which can be
used at the receiver to overcome the effects of noise and interference encountered in the
transmission of the signal through the channel. Thus, the added redundancy serves to increase
the reliability of the received data and improves the fidelity of the received signal. In effect,
redundancy in the information sequence aids the receiver in decoding the desired information
sequence. For example, a (trivial) form of encoding of the binary information sequence is
simply to repeat each binary digit m times, where m is some positive integer. More
sophisticated (nontrivial) encoding involves taking k information bits at a time and mapping
each k-bit sequence into a unique n-bit sequence, called a code word. The amount of
redundancy introduced by encoding the data in this manner is measured by the ratio n/k. The
reciprocal of this ratio, namely, k/n is called the rate of the code or, simply, the code rate.
The binary sequence at the output of the channel encoder is passed to the digital modulator,
which serves as the interface to the communications channel. Since nearly all of the
communication channels encountered in practice are capable of transmitting electrical signals
(waveforms), the primary purpose of the digital modulator is to map the binary information
sequence into signal waveforms. To elaborate on this point, let us suppose that the coded
information sequence is to be transmitted one bit at a time at some uniform rate R bits/s. The
digital modulator may simply map the binary digit 0 into a waveform s0(t) and the binary digit
1 into a waveform s1(t). In this manner, each bit from the channel encoder is transmitted
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separately. We call this binary modulation. Alternatively, the modulator may transmit b coded
information bits at a time by using M =2b distinct waveform si(t), I = 0, 1, …, m-1, one
waveform for each of the 2b possible b-bits sequences. We call this M-ary modulation (M
>2). Note that a new b-bit sequence enters the modulator every b/R seconds. Hence, when the
channel bit rate R is fixed, the amount of time available to transmit one of the M waveforms
corresponding to a b-bit sequence is b times the period in a system that uses binary
modulation.
At the receiving end of a digital communications system, the digital demodulator processes
the channel-corrupted transmitted waveform and reduces each waveform to a single number
that represents an estimate of the transmitted data symbol. For example, when binary
modulation is used, the demodulator may process the received waveform and decide on
whether the transmitted bit is a 0 or 1. In such a case, we say the demodulator has made a
binary decision. As one alternative, the demodulator may make a ternary decision; that is, it
decides that the transmitted bit is either a 0 or 1 or it makes no decision at all, depending on
the apparent quality of the received signal. When no decision is made on a particular bit, we
say that the demodulator has inserted an erasure in the demodulated data. Using the
redundancy in the transmitted data, the decoder attempts to fill in the positions where erasures
occurred. Viewing the decision process performed by the demodulator as a form of
quantization, we observe that binary and ternary decisions are special cases of a demodulator
that quantizes to Q levels, where Q ≥ 2 In general, if the digital communications system
employs M-ary modulation, where m = 0,1, … , M represent the M possible transmitted sym-
bols, each corresponding to k =log2 M bits, the demodulator may make A Q-ary decision,
where Q ≥ M .In the extreme case where no quantization is performed, Q = ∞.
When there is no redundancy in the transmitted information, the demodulator must decide
which of the M waveforms was transmitted in any given time interval. Consequently, Q = M,
and since there is no redundancy in the transmitted information, no discrete channel decoder
is used following the demodulator. On the other hand, when there is redundancy introduced
by a discrete channel encoder at the transmitter, the Q-ary output from the demodulator
occurring every k/R seconds is fed to the decoder, which attempts to reconstruct the original
information sequence from knowledge of the code used by the channel encoder and the
redundancy contained in the received data. A measure of how well the demodulator and
encoder perform is the frequency with which errors occur in the decoded sequence. More
precisely, the average probability of a bit-error at the output of the decoder is a measure of the
performance of the demodulator-decoder combination. In general, the probability of error is a
function of the code characteristics, the types of waveforms used to transmit the information
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over the channel, the transmitter power, the characteristics of the channel (i.e., the amount of
noise), the nature of the interference, etc., and the method of demodulation and decoding.
These items and their effect on performance will be discussed in detail in subsequent
chapters.
As a final step, when an analog output is desired, the source decoder accepts the output
sequence from the channel decoder, and from knowledge of the source encoding method
used, attempts to reconstruct the original signal from the source. Due to channel decoding
errors and possible distortion introduced by the source encoder and, perhaps, the source
decoder, the signal at the output of the source decoder is an approximation to the original
source output. The difference or some function of the difference between the original signal
and the reconstructed signal is a measure of the distortion introduced by the digital
communications system.
Early Work in Digital Communications

Although Morse is responsible for the development of the first electrical digital
communication system (telegraphy), the beginnings of what we now regard as modem digital
communications stem from the work of Nyquist (1924), who investigated the problem of
determining the maximum signalling rate that can be used over a telegraph channel of a given
bandwidth without intersymbol interference. He formulated a model of a telegraph system in
which a transmitted signal has the general form
S (t )   a n g (t  nT )
n
Where g(t) represents a basic pulse shape and {an} is the binary data sequence of {±1}
transmitted at a rate of 1/T bits per second. Nyquist set out to determine the optimum pulse
shape that was bandlimited to W Hz and maximised the bit rate 1/T under the constraint that
the pulse caused no intersymbol interference at the sampling times k/T, k = 0, ±1, ±2,.... His
studies led him to conclude that the maximum pulse rate1/T is 2W pulses per second. This rate
is now called the Nyquist rate. Moreover, this pulse rate can be achieved by using the pulses
g(t) = (sin2π Wt)/2π Wt. This pulse shape allows the recovery of the data without intersymbol
interference at the sampling instants Nyquist’s result is equivalent to a version of the sampling
theorem for bandlimited signals, which was later stated precisely by Shannon (1948). The
sampling theorem states that a signal of bandwidth W can be reconstructed from samples
taken at the Nyquist rate of 2W samples per second using the interpolation formula
n sin 2W (t  n / 2W )
S (t )   s( )
n 2W 2W (t  n / 2W )
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In light of Nyquist's work Hartley (1928) considered the issue of the amount of data that can
be transmitted reliably over a bandlimited channel when multiple amplitude levels are used.
Due to the presence of noise and other interference, Hartley postulated that the receiver could
reliably estimate the received signal amplitude to some accuracy, say Aδ. This investigation
led Hartley to conclude that there is maximum data rate that can be communicated reliably
over a bandlimited channel when the maximum signal amplitude is limited to Amax (fixed
power constraint) and the amplitude resolution is Aδ.
Another significant advance in the development of communications was the work of Wiener
(1942) who considered the problem of estimating a desired signal waveform s(t) in the
presence of additive noise n(t), based on observation of the received signal r(t) = s(t) + n(t).
This problem arises in signal demodulation. Wiener determined the linear filter whose output
is the best mean-square approximation to the desired signal s(t). The resulting filter is called
the optimum linear (Wiener) filter. Hartley’s and Nyquist results on the maximum
transmission rate of digital information were precursors to the work of Shannon (1948 a, b)
who established the mathematical foundations for information theory and derived the
fundamental limits for digital communication systems. In his pioneering work, Shannon
formulated the basic problem of reliable transmission of information in statistical terms, using
probabilistic models for information sources and communication channels. Based on such a
statistical formulation, he adopted a logarithmic measure for the information content of a
source. He also demonstrated that the effect of a transmitter power constraint, a bandwidth
constraint, and additive noise can be associated with the channel and incorporated into a
single parameter, called the channel capacity For example, in the case of an additive white
(spectrally flat) Gaussian noise interference, an ideal bandlimited channel of bandwidth W has
a capacity C given by
P
C = W log2(1 + ———) bits/s
WN0
where P is the average transmitted power and N0 is the power spectral density of the additive
noise. The significance of the channel capacity is as follows: If the information rate R from
the source is less than C (R < C), then it is theoretically possible to achieve reliable (error-
free) transmission through the channel by appropriate coding. On the other hand, if R > C,
reliable transmission is not possible regardless of the amount of signal processing performed
at the transmitter and receiver. Thus, Shannon established basic limits on communication of
information and gave birth to a new field that is now called information theory.
Initially the fundamental work of Shannon had a relatively small impact on the design and
development of new digital communications systems. In part, this was due to the small
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demand for digital information transmission during the 1950's. Another reason was the
relatively large complexity and, hence, the high cost of digital hardware required to achieve
the high efficiency and high reliability predicted by Shannon's theory.
Another important contribution to the field of digital communications is the work of
Kotelnikov (1947), which provided a coherent analysis of the various digital communication
systems based on a geometrical approach. Kotelnikov approach was later expanded by
Wozencraft and Jacobs (1965).
The increase in the demand for data transmission during the last three decades, coupled with
the development of more sophisticated integrated circuits, has led to the development of very
efficient and more reliable digital communications systems. In the course of these
developments, Shannon's original results and the generalization of his results on maximum
transmission limits over a channel and on bounds on the performance achieved have served as
benchmarks for any given communications system design. The theoretical limits derived by
Shannon and other researchers that contributed to the development of information theory
serve as an ultimate goal in the continuing efforts to design and develop more efficient digital
communications systems.
Following Shannon's publications name the classic work of Hamming (1950) on error
detecting and error-correcting codes to combat the detrimental effects of channel noise.
Hamming's work stimulated many researchers in the years that followed, and a variety of new
and powerful codes were discovered, many of which are used today in the implementation of
modem communication systems.
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CHAPTER II
DATA TRANSMISSION & SIGNALS
Data Transmission
Concepts and Terminology
 Transmission Terminology
Transmission from transmitter to receiver goes over some transmission medium using
electromagnetic waves.
- Guided Media: waves are guided along a physical path; twisted pair, optical fibre,
coaxial cable.
- Unguided Media: waves are not guided; air waves radio waves.
- Direct Link: signal goes from transmitter to receiver without intermediate devices,
other than amplifiers and repeaters.
- Point-to Point Link: guided media with direct link between two devices.
- Multipoint Guided Configuration: more than two devices can share the same
medium.
 Frequency, Spectrum, & Bandwidth

- Signal is generated by a transmitter and transmitted over a medium.
- Signal is a function of time or frequency.
A signal is any function that carries information. Based on the range of variation of
independent variables, signals can be divided into two classes: continuous-time (or
analog) signals and discrete-time (or digital) signals. A signal is a function of time, but
can also be expressed as a function of frequency; that is, the signal consists of components
of different frequencies.
 Analog and Digital Signals

Information can be analog or digital. Analog information is continuous. Digital
information is discrete.
Signals can be analog or digital. Analog signals can have any value in a range; while
digital signals can have only a limited number of values.
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Value Value
Analog Signal Digital Signal
 Time –Domain Concepts

- Continuous Signal: Signal intensity varies in a smooth fashion over time; no breaks
or discontinuities in the signal.
- Discrete Signals: Signal intensity can take one of two pre-specified values for any
amount of time.
A continuous time signal is defined by a continuous independent variable. A signal s(t) is
continuous if
lim sc (t )  s(t ) for all a.

t a
- Periodic Signal
A signal s(t) is periodic if
s(t  T )  s(t )
where T is the period of the signal.
sc(t) sd(t)
A
t t
 T
T
Three important characteristics of a periodic signal are: amplitude, frequency, and phase.
Amplitude (A) is the instantaneous value of a signal at any time, and is measured in volts.
Frequency (f) is the inverse of the period (T); (f=1/T), or the number of period repetition in
one second, and is measured in cycles per second or Hertz (Hz). Phase ( ) is a measure of
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the relative position in time within a single period of a signal. Thus, we can express a
sinusoid signal as
s(t )  A sin(2ft   )
where A is the amplitude, f is the frequency, and  is the phase.
 Frequency-Domain Concepts
Any signal can also be viewed as a function of frequency, for example, the signal
s(t )  sin 2ft  1 / 3 sin 3(2ft )  1 / 5 sin 5(2ft )

consists of three components as shown in the figure below:
sin 2ft
1/3 sin 3(2ft)
1/5 sin 5(2ft)
s(t)
The frequency components of a signal can be determined using Fourier analysis. The
following figure shows the spectrum S(f) of the signal s(t). The spectrum of a signal is the
range of frequencies that it contains. For this signal the spectrum extends from f to 5f . the
spectrum in this case is discrete.
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S(f)
1
1/3
1/5
f
f 3f 5f
Many signals, such as the one in the following figure, have continuous spectrum Sc(f) and
sd(t)
t
T
and an infinite bandwidth as shown below:

Sc(f)
1/T 2/T 3/T 4/T n/T

f
However, most of the energy in the signal is contained in a relatively narrow band of
frequencies. This band is referred to as the effective bandwidth, or just bandwidth.
If a signal includes a component of zero frequency, that component is called dc component
or constant component.
The signal s1(t) in the following figure is obtained by adding a dc component on s(t):
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s1(t)
With a dc component, it has a frequency term at f = 0 and a non-zero average amplitude.
S1(f)
1.2 1
Spectrum of s1(t)
1/3
1/5
0 f 3f 5f f
s1 (t )  1.2  sin (2ft )  1 / 3 sin 3(2ft )  1 / 5 sin 5(2ft )
 Fundamental Frequency
Base frequency such that the frequency of all components can be expressed as its integer
multiples; the period of the aggregate signal is the same as the period of the fundamental
frequency:
- Each signal can be decomposed into a set of sinusoid signals by making use of
Fourier’s analysis.
- The time-domain function s(t) specifies a signal in terms of its amplitude at each
instant of time.
- The frequency-domain function S(f) specifies the signal in terms of peak amplitude of
constituent frequencies.
Spectrum
Range of frequencies contained in a signal.
Absolute Bandwidth
Width of the spectrum.
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Effective Bandwidth
Narrow band of frequencies containing most of the energy of the signal.
DC Component
Component of zero frequency; changes the average amplitude of the signal to non-zero.
Relationship between Data Rate and Bandwidth

- Any transmitter/receiver system can accommodate only a limited range of frequencies.
* The range for FM radio transmission is 88-108 MHz
- This limits the data rate that can be carried over the transmission medium.
- Consider a square wave. Suppose that we let the positive pulse to be binary 1 and the
negative pulse to be binary 0. Then, the waveform represents the binary stream
1010… and duration (period) of each pulse is 1/2f. Thus, the data rate is equal to 2f
bits per second (bps) or the data rate is equal to twice the fundamental frequency of
the digital signal. It can be shown that the frequency-domain representation of this
waveform is:

1
s(t )   sin(2kft )
k 1 k
- This waveform has infinite number of frequency components and infinite bandwidth.
- Peak amplitude of the kth frequency component is 1/k, so most of the energy is
concentrated in the first few frequencies.
Ex
Consider a digital transmission system capable of transmitting signals with a bandwidth of
4 MHz.
Case 1
Approximate the square wave with a waveform of the first three sinusoidal components
sin (2ft )  1 / 3 sin(2 (3 f )t )  1 / 5 sin(2 (5 f )t )

If f = 106 cycles per second, or 1 MHz, the bandwidth of the signal

s(t )  sin (2  10 6 t )  1 / 3 sin(2  3  10 6 t )  1 / 5 sin(2  5  10 6 t ) 
is 5x106-106 = 4 MHz
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For f = 1 MHz, the period of the fundamental frequency is T = 1/10 6 = 1s. If the
waveform is a bit string of 1’s and 0’s, then one bit occurs every 0.5 s for a data rate of
2x106 bps or 2 Mbps.
Case 2
Assume a bandwidth of 8 MHz and f = 2 MHz; this gives us the signal bandwidth as
(5x2x106)-(2x106) = 8 MHz
But T = 1/f = 0.5 s, so that the time needed for one bit is 0.25 s, giving a data rate of
4 Mbps. Other things being equal, doubling the bandwidth doubles the potential data rate.
Case 3
Let us represent the signal by the first two components of the sinusoid as
sin (2ft )  1/ 3sin(2 (3 f )t )
Assume that f = 2 MHz and T = 1/f = 0.5 s so that the time needed for one bit is 0.25 s,
giving a data rate of 4 Mbps.
Bandwidth of the signal is
(3x2x106)-(2x106) = 4 MHz
A given bandwidth can support various data rates depending on the ability of the receiver
to differentiate between 0 and 1 in the presence of noise and other impairments.
Ex
If a periodic signal is decomposed into five waves with frequencies of 100, 300, 500, 700,
and 900 Hz, what is the bandwidth of the signal?
Let fh be the highest frequency, fl be the lowest frequency, and B be the bandwidth, then
B = fh - fl
= 900-100 = 800 Hz
Digital Signals
Data can be represented by a digital signal. For example, a 1 can be encoded as a positive
voltage and a 0 as a zero voltage.
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Amplitude
1 0 1 1 0 0 0 1
…
Time
Amplitude, Period and Phase

The three characteristics of periodic signals can be redefined for a periodic digital signal
Amplitude …
Time
No phase shift
Amplitude …
Time
180o phase shift
Amplitude Amplitude
Time Time
¼ cycle
0o phase shift 90o phase shift
(no phase shift)
Amplitude Amplitude
Time Time
½ cycle
180o phase shift 270o phase shift
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Bit Interval and Bit Rate

Most digital signals are aperiodic and thus terms like period or frequency are not
appropriate. Two new terms, bit interval (instead of period) and bit rate (instead of
frequency) are used to describe a digital signal. The bit interval is the time required to
send one single bit. The bit rate is the number of bit intervals per second. This means that
the bit rate is the number of bits sent in one second, usually expressed in bits per second
(bps).
Amplitude
1 second = 8 bit intervals
bit rate = 8 bps
1 0 1 1 0 0 0 1
…
Time
bit interval
Decomposition of a Digital Signal

A digital signal can be decomposed into an infinite number of simple sine waves called
harmonics, each with different amplitude, frequency and phase. This means that when a
digital signal is sent along a transmission medium, an infinite number of simple signals is
being sent.
Harmonics of a Digital Signal
… …
a) First harmonic only b) First, third, and fifth harmonics
… …
c) First, third, fifth, and seventh harmonics d) Infinite number of harmonics

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If some of the components do not pass through the medium, this results in distortion of
the signal at the receiver side. Since no practical medium (such as a cable) is capable of
transferring the entire range of frequencies, there will always be distortion.
Amplitude
Frequency
0 Infinite bandwidth Infinity
a) Spectrum for exact replica
Amplitude
Frequency
Significant bandwidth
b) Significant spectrum
The part of the infinite spectrum whose amplitudes are significant (above an acceptable
threshold), is called the significant spectrum, and its bandwidth is called the significant
bandwidth.
When the bit rate increases, the significant bandwidth widens. For example, if the bit rate
is 1000 bps, the significant bandwidth can be around 200 Hz, depending on the level of
noise in the system. If the bit rate is 2000 bps, the significant bandwidth can be 400 Hz.
1000 bps
200 Hz
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2000 bps
400 Hz
A transmission medium with a particular bandwidth is capable of transmitting only digital
signals whose significant bandwidth is less than the bandwidth of the medium.
Channel Capacity
The maximum bit rate a transmission medium can transfer is called channel capacity of
the medium. The capacity of a channel, however, depends on the type of encoding
technique and the signal-to-noise ratio of the system. For example a normal telephone line
with a bandwidth of 3000 Hz is capable of transferring up to 20,000 bps, but other factors,
like noise, can decrease this rate.
1000 bps
Bandwidth = x
2000 bps
Bandwidth = 2x
3000 bps
Bandwidth = 3x
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Noise
In the absence of a signal, a transmission medium ideally has no electrical signal present.
In practice, however, there is what we call line noise level, because of random
perturbations on the line even when no signal is being transmitted. An important
parameter associated with a transmission medium, therefore, is the ratio of the average
power in a received signal, S , to the power in the noise level, N . The ratio S / N is
known as the signal-to-noise ratio (SNR) and normally is expressed in decibels, that is:
S
SNR = 10 log 10  dB
N
- A high SNR ratio means a good-quality signal.
- A low SNR ratio means a low-quality signal.
The theoretical maximum data rate of transmission channel is related to the SNR ratio
and we can determine this rate using a formula attributed to Shannon and Hartley. This is
known as the Shannon-Hartley Law, which states:
 S
C  W log 2 1   bps
 N
 S
 3.32 W log 10 1   bps
 N
where C is the data rate in bps, W is the bandwidth of the line channel in Hz, S is the
average signal power in watts and N is the random noise power in watts.
Ex
Consider a voice channel with BW of 2,800 Hz. A typical value of S/N for a telephone
line is 20 dB. What is the channel capacity?
Solution
SNR = 20 dB
20 = 10 log10 (S/N)  S/N = 100
W = 2,800 Hz
 S  S
C  W log 2 1   bps  3.32 W log 10 1   bps
 N  N
 3.32 (2800) log10 1  100
C = 18,632 bps
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CHAPTER 3
TRANSMISSION MEDIA
There are two basic categories of transmission media: guided and unguided media.
Guided transmission media use cabling system that guides the data signals along a specific
path. Data signals are bound by the cabling system. Guided media is also known as bound
media. ―Cabling‖ is meant in a generic sense, and is not meant to be interpreted as copper
wire cabling only.
Unguided transmission media consists of a means for the data signals to travel but nothing
to guide them along a specific path. The data signals are not bound to a cabling media and are
therefore often called unbound media.
Transmission Media: Guided
There four basic types of guided media:
a. Open Wire
b. Twisted Pair
c. Coaxial Cable
d. Optical Fibre
Open Wire
Open wire is traditionally used to describe the electrical wire strung along power poles. There
is a single wire strung between poles. No shielding or protection from noise interference is
used. We are going to extend the traditional definition of open wire to include any data signal
path without shielding or protection from noise interference. This can include multi conductor
cables or single wires. This medium is susceptible to a large degree of noise and interference
and consequently is not acceptable for data transmission except for short distances under
20 ft.
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Twisted Pair
The wires in twisted pair cabling are twisted together in pairs. Each pair consists of a wire
used for the positive data signal and a wire used for the negative data signal. Any noise that
appears on one wire of the pair will also occur on the other wire. Since the wires have
opposite polarities, they are 180 degrees out of phase. When noise appears on both wires, it
cancels or nulls itself out at the receiving end. Twisted pair cables are most effectively used in
systems that use a balanced line method of transmission: polar line coding (Manchester
encoding) as opposed to unipolar line coding.
Unshielded Twisted Pair
The degree of reduction in noise interference is determined specifically by the number of

turns per foot. Increasing the number of turns per foot reduces the noise interference. To
further improve noise rejection, a foil or wire braid ―shield‖ is woven around the twisted
pairs. This shield can be woven around individual pairs or around a multi-pair conductor
(several pairs).
Shielded Twisted Pair
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Cables with a shield are called shielded twisted pair and are commonly abbreviated STP.
Cables without a shield are called unshielded twisted pair or UTP. Twisting the wires together
results in a characteristic impedance for the cable. Typical impedance for UTP is 100 Ohm for
Ethernet 10BaseT cable.
UTP or unshielded twisted pair cable is used in Ethernet 10BaseT and can also be used with
Token Ring. It uses the RJ line of connectors (RJ45, RJ11, etc..).
STP or shielded twisted pair is used with the traditional Token Ring cabling or ICS-IBM
Cabling System. It requires a custom connector. IBM STP (shielded twisted pair) has a
characteristic impedance of 150 Ohm.
Coaxial Cable
Coaxial cable consists of two conductors. The inner conductor is held inside an insulator with
the other conductor woven around it providing a shield. An insulating protective coating
called a jacket covers the outer conductor.
Coaxial Cable
The outer shield protects the inner conductor from outside electrical signals. The distance
between the outer conductor (shield) and inner conductor, plus the type of material used for
insulating the inner conductor determine the cable properties or impedance. Typical
impedances for coaxial cables are 75 Ohms for TV cable, 50 Ohms for Ethernet Thinnet and
Thicknet. The excellent control of the impedance characteristics of the cable allow higher data
rates to be transferred than with twisted pair cable.
Optical fibre
Optical fibre consists of thin glass fibres that can carry information at frequencies in the
visible light spectrum and beyond. The typical optical fibre consists of a very narrow strand of
glass called the core. Around the core is a concentric layer of glass called the cladding. A
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typical core diameter is 62.5 microns (1 micron = 10-6 m). Typically Cladding has a diameter
of 125 microns. Coating the cladding is a protective coating consisting of plastic, it is called
the Jacket.
Fibre Optic Cables
Just as standard electric cables come in a variety of sizes, shapes, and types, fibre optic cables
are available in different configurations. The simplest cable is just a single strand of fibre,
whereas complex cables are made up of multiple fibres with different layers and other
elements.
The portion of a fibre optic cable (core) that carries the light is made from either glass or
plastic. Another name for glass is silica. Special techniques have been developed to create
nearly perfect optical glass or plastic, which is transparent to light. Such materials can carry
light over a long distance. Glass has superior optical characteristics over plastic. However,
glass is far more expensive and more fragile than plastic. Although the plastic is less
expensive and more flexible, its attenuation of light is greater. For a given intensity, light will
travel a greater distance in glass than in plastic. For very long distance transmission, glass is
certainly preferred. For shorter distances, plastic is much more practical.
All fibres consist of a number of substructures including:
A core, which carries most of the light, surrounded by

A cladding, which bends the light and confines it to the core, surrounded by
A substrate layer (in some fibres) of glass which does not carry light, but adds to the diameter
and strength of the fibre, covered by
A primary buffer coating, this provides the first layer of mechanical protection, covered by
A secondary buffer coating, this protects the relatively fragile primary coating.
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The cladding is also made of glass or plastic but has a lower index of refraction. This ensures
that the proper interface is achieved so that the light waves remain within the core. In addition
to protecting the fibre core from nicks and scratches, the cladding adds strength. Some fibre
optic cables have a glass core with a glass cladding. Others have a plastic core with a plastic
cladding. Another common arrangement is a glass core with a plastic cladding. It is called
plastic-clad silica (PCS) cable.
An important characteristic of fibre optics is refraction. Refraction is the characteristic of a
material to either pass or reflect light. When light passes through a medium, it "bends" as it
passes from one medium to the other. An example of this is when we look into a pond of
water.
In 1621, the Dutch mathematician Willebrard Snell established that rays of light can be traced
as they propagate from one medium to another based on their indices of refraction. Snell’s
low is stated by the equation:
Incident ray
Reflected ray
1 
Air
Water
2 Refracted ray
n1 sin θ 2
 ;
n2 sin θ1
n1 sin 1 = n2 sin 2
where n1-refractive index of material 1; 1-angle of incidence; 2-angle of refraction;

n2-refractive index of material 2. When the angle of incidence, 1, becomes large enough to
cause the sine of the refraction angle, 2, to exceed the value of 1, total internal reflection
occurs. This angle is called the critical angle, c. The critical angle, c, can be derived from
Snell’s law as follows
n1 sin 1 = n2 sin 2
sin 1 = n2 sin 2/n1
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When sin 1 = sin 2, then sin 1 = n2 / n1. Therefore, the critical angle: c = sin -1 (n2 / n1)
Its index of refraction, however, it is typically 1% less than that of its core. This permits total
internal reflection of rays entering the fibre and striking the core-cladding interface above the
critical angle of approximately 82-degree (sin-1 (1/1.01). The core of the fibre therefore guides
the light and the cladding contains the light. The cladding material is much less transparent
than the glass making up the core of the fibre. This causes light rays to be absorbed if they
strike the core-cladding interface at an angle less than the critical angle.
If the angle of incidence is small, the light rays are reflected and do not pass into the water. If
the angle of incident is great, light passes through the media but is bent or refracted.
In the following figure, a light ray is transmitted into the core of an optical fibre. Total
Cladding
Core
Figure 2
internal reflection occurs as it strikes the lower index cladding material.
Optical fibres work on the principle that the core refracts the light and the cladding reflects
the light. The core refracts the light and guides the light along its path. The cladding reflects
any light back into the core and stops light from escaping through it.
Transmission Modes
There are three primary types of transmission modes using optical fibre. They are
a. Step Index
b. Graded Index
c. Single Mode
Step index has a large core, so the light rays tend to bounce around inside the core, reflecting
off the cladding. This causes some rays to take a longer or shorter path through the core.
Some take the direct path with hardly any reflections while others bounce back and forth
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taking a longer path. The result is that the light rays arrive at the receiver at different times.
The signal becomes longer than the original signal. LED light sources are used. Typical Core:
62.5 microns.
Graded index has a gradual change in the core's refractive index. This causes the light rays to
be gradually bent back into the core path. This is represented by a curved reflective path in the
attached drawing. The result is a better receive signal than with step index. LED light sources
are used. Typical Core: 62.5 microns.
Note: Both step index and graded index allow more than one light source to be used (different
colours simultaneously), so multiple channels of data can be run at the same time!
Single mode has separate distinct refractive indexes for the cladding and core. The light ray
passes through the core with relatively few reflections off the cladding. Single mode is used
for a single source of light (one colour) operation. It requires a laser and the core is very
small: 9 microns.
Basic Construction of Fibre-Optic Cables

There are two basic ways of classifying fibre optic cables. The first way is an indication of
how the index of refraction varies across the cross section of the cable. The second way of
classification is by mode. Mode refers to the various paths that the light rays can take in
passing through the fibre. Usually these two methods of classification are combined to define
the types of cable. There are two basic ways of defining the index of refraction variation
across a cable. These are step index and graded index. Step index refers to the fact that there
is a sharply defined step in the index of refraction where the fibre core and the cladding
interface. It means that the core has one constant index of refraction n1, while the cladding has
another constant index of refraction n2.
The other type of cable has a graded index. In this type of cable, the index of refraction of the
core is not constant. Instead, the index of refraction varies smoothly and continuously over the
diameter of the core. As you get closer to the centre of the core, the index of refraction
gradually increases, reaching a peak at the centre and then declining as the other outer edge of
the core is reached. The index of refraction of the cladding is constant.
Mode refers to the number of paths for the light rays in the cable. There are two
classifications: single mode and multimode. In single mode, light follows a single path
through the core. In multimode, the light takes many paths through the core.
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Each type of fibre optic cable is classified by one of these methods of rating the index or
mode. In practice, there are three commonly used types of fibre optic cable: multimode step
index, single mode step index and multimode graded index cables.
1. Multimode Step-Index Fibre.

This cable (see Figure 5 (a)) is the most common and widely used type. It is also the easiest to
make and, therefore, the least expensive. It is widely used for short to medium distances at
relatively low pulse frequencies.
Cross section Index profile Beam path dispersion
n1
n2
C B
A
cladding a)
Input
core Input
Output
Light Output
source b)
Figure 5 c)
The main advantage of a multimode step index fibre is the large size. Typical core diameters
are in the 50-to-1000 micrometers (m) range. Such large diameter cores are excellent at
gathering light and transmitting it efficiently. This means that an inexpensive light source
such as LED can be used to produce the light pulses. The light takes many hundreds of even
thousands of paths through the core before exiting. Because of the different lengths of these
paths, some of the light rays take longer to reach the end of the cable than others. The
problem with this is that it stretches the light pulses (Figure 5 (b). In Figure 5 ray A reaches
the end first, then B, and C. The result is a pulse at the other end of the cable that is lower in
amplitude due to the attenuation of the light in the cable and increased in duration due to the
different arrival times of the various light rays. The stretching of the pulse is referred to as
modal dispersion. Because the pulse has been stretched, input pulses can not occur at a rate
faster than the output pulse duration permits. Otherwise the pulses will essentially merge
together as shown in Figure 5 (c). At the output, one long pulse will occur and will be
indistinguishable from the three separate pulses originally transmitted. This means that
incorrect information will be received. The only core for this problem is to reduce the pulse
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repetition rate. When this is done, proper operation occurs. But with pulses at a lower
frequency, less information can be handled.
2. Single Mode Cable

In a single mode, or mono-mode, step-index fibre cable the core is so small that the total
number modes or paths through the core are minimised and modal dispersion is essentially
eliminated. The typical core sizes are 5 to 15 m. The output pulse has essentially the same
duration as the input pulse (see Figure 6).
The single mode step index fibres are by far the best since the pulse repetition rate can be high
and the maximum amount of information can be carried. For very long distance transmission
and maximum information content, single-mode step-index fibre cables should be used.
The main problem with this type of cable is that because of its extremely small size, it is
difficult to make and is, therefore, very expensive. Handling, splicing, and making
interconnections are also more difficult. Finally, for proper operation an expensive, super
intense light source such as a laser must be used. For long distances, however, this is the type
of cable preferred.
Cross section Index profile Beam path

n1
n2
cladding
Input Output
core
Figure 6
3. Multimode graded-index fibre cables

These cables have a several modes or paths of transmission through the cable, but they are
much more orderly and predictable. Figure 7 shows the typical paths of the light beams.
Because of the continuously varying index of refraction across the core, the light rays are bent
smoothly and converge repeatedly at points along the cable.
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Cross section Index profile Beam path

n1
n2
cladding
Input Output
core
Figure 7
The light rays near the edge of the core take a longer path but travel faster since the index of
refraction is lower. All the modes or light paths tend to arrive at one point simultaneously.
The result is that there is less modal dispersion. It is not eliminated entirely, but the output
pulse is not nearly as stretched as in multimode step index cable. The output pulse is only
slightly elongated. As a result, this cable can be used at very high pulse rates and, therefore, a
considerable amount of information can be carried on it.
This type of cable is also much wider in diameter with core sizes in the 50 to 100 (m) range.
Therefore, it is easier to splice and interconnect, cheaper, and less-intense light sources may
be used. The most popular fibre-optic cables that are used in LAN: multimode-step index
cable -65.5/125; multimode-graded index cable - 50/125. The multimode-graded index cable -
100/140 or 200/300 are recommended for industrial control applications because of its large
size. In high data rate systems single mode fibre 9/125 is used. Typical core and cladding
diameters of these cables are shown in Figure 8.
9
62.5
140
125
50
100
125
125
Figure 8
Specifications of the Fibre Cables
Indoor cable specifications:
 LED (Light Emitting Diode) light source

 3.5 dB/Km Attenuation (loses 3.5 dB of signal per kilometer)
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 850 nM - wavelength of light source
 Typically 62.5/125 (core diameter/cladding diameter)
 Multimode - can run many light sources.
Outdoor cable specifications:
 Laser Light Source

 1 dB/Km Attenuation (loses 1 dB of signal per kilometer)
 1170 nM - wavelength of light source
 Monomode (single mode)
Advantages of Optical Fibre:
 Noise immunity: RFI and EMI immune (RFI - Radio Frequency Interference, EMI -
Electromagnetic Interference)
 Security: cannot tap into cable.
 Large Capacity due to BW (bandwidth)
 No corrosion
 Longer distances than copper wire
 Smaller and lighter than copper wire
 Faster transmission rate
Disadvantages of optical fibre:
 Physical vibration will show up as signal noise!

 Limited physical arc of cable. Bend it too much and it will break!
 Difficult to splice
The cost of optical fibre is a trade-off between capacity and cost. At higher transmission
capacity, it is cheaper than copper. At lower transmission capacity, it is more expensive.
Media versus Bandwidth
The following table compares the usable bandwidth of the different guided transmission
media.
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Cable Type Bandwidth
Open Cable 0 - 5 MHz
Twisted Pair 0 - 100 MHz
Coaxial Cable 0 - 600 MHz
Optical Fibre 0 - 1 GHz
Transmission Media: Unguided
Unguided transmission media is data signals that flow through the air. They are not guided or
bound to a channel to follow. They are classified by the type of wave propagation.
RF Propagation
There are three types of RF (radio frequency) propagation:
 Ground Wave
 Sky Wave
 Line of Sight (LOS)
Ground wave propagation follows the curvature of the Earth. Ground waves have carrier
frequencies up to 2 MHz. AM radio is an example of ground wave propagation.
Sky wave propagation bounces off of the Earth's ionospheric layer in the upper atmosphere.
It is sometimes called double hop propagation. It operates in the frequency range of 30-85
MHz. Because it depends on the Earth's ionosphere, it changes with the weather and time of
day. The signal bounces off of the ionosphere and back to earth. Ham radios operate in this
range.
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Line of sight propagation transmits exactly in the line of sight. The receive station must be
in the view of the transmit station. It is sometimes called space waves or troposphere
propagation. It is limited by the curvature of the Earth for ground-based stations (100 km,
from horizon to horizon). Reflected waves can cause problems. Examples of line of sight
propagation are: FM radio, microwave and satellite.
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Radio Frequencies
The frequency spectrum operates from 0 Hz (DC) to gamma rays (1019 Hz).
Name Frequency (Hertz) Examples
Gamma Rays 1019
X-Rays 1017
Ultra-Violet Light 7.5 x 1015
Visible Light 4.3 x 1014
Infrared Light 3 x 1011
EHF - Extremely High Frequencies 30 GHz (Giga = 109) Radar
SHF - Super High Frequencies 3 GHz Satellite & Microwaves
UHF - Ultra High Frequencies 300 MHz (Mega = 106) UHF TV (Ch. 14-83)
VHF - Very High Frequencies 30 MHz FM & TV (Ch2 - 13)
HF - High Frequencies 3 MHz2 Short Wave Radio
MF - Medium Frequencies 300 kHz (kilo = 103) AM Radio
LF – Low Frequencies 30 kHz Navigation
VLF - Very Low Frequencies 3 kHz Submarine Communications
VF - Voice Frequencies 300 Hz Audio
ELF - Extremely Low Frequencies 30 Hz Power Transmission
Radio frequencies are in the range of 300 kHz to 10 GHz. We are seeing an emerging
technology called wireless LANs. Some use radio frequencies to connect the workstations
together, some use infrared technology.
Microwave
Microwave transmission is line of sight transmission. The transmit station must be in visible
contact with the receive station. This sets a limit on the distance between stations depending
on the local geography. Typically the line of sight due to the Earth's curvature is only 100 km
to the horizon! Repeater stations must be placed so the data signal can hop, skip and jump
across the country.
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Microwaves operate at high operating frequencies of 3 to 10 GHz. This allows them to carry
large quantities of data due to their large bandwidth.
Advantages:
a. They require no right of way acquisition between towers.

b. They can carry high quantities of information due to their high operating frequencies.
c. Low cost land purchase: each tower occupies only a small area.
d. High frequency/short wavelength signals require small antennae.
Disadvantages:
a. Attenuation by solid objects: birds, rain, snow and fog.

b. Reflected from flat surfaces like water and metal.
c. Diffracted (split) around solid objects.
d. Refracted by atmosphere, thus causing beam to be projected away from receiver.
Satellite
Satellites are transponders (units that receive on one frequency and retransmit on another) that
are set in geostationary orbits directly over the equator. These geostationary orbits are 36,000
km from the Earth's surface. At this point, the gravitational pull of the Earth and the
centrifugal force of Earth's rotation are balanced and cancel each other out. Centrifugal force
is the rotational force placed on the satellite that wants to fling it out into space.
The uplink is the transmitter of data to the satellite. The downlink is the receiver of data.
Uplinks and downlinks are also called Earth stations because they are located on the Earth.
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The footprint is the "shadow" that the satellite can transmit to, the shadow being the area that
can receive the satellite's transmitted signal.
Iridium Telecom System
The Iridium Telecom System is a new satellite system that will be the largest private
aerospace project. It is a mobile telecom system intended to compete with cellular phones. It
relies on satellites in lower Earth orbit (LEO). The satellites will orbit at an altitude of 900 -
10,000 km in a polar, non-stationary orbit. Sixty-six satellites are planned. The user's handset
will require less power and will be cheaper than cellular phones. There will be 100% coverage
of the Earth.
Unfortunately, although the Iridium project was planned for 1996-1998, with 1.5 million
subscribers by end of the decade, it looked very financially unstable.
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CHAPTER 4
ENCODING, MODULATING & TRANSMISSION CODES
ENCODING (D/D) (A/D) (D/A) (A/A)

Information must be encoded into signals before it can be transported across
communication media. We must encode data into signals to send them from one place
to another.
Digital-to-Digital Encoding
Digital-to-Digital Encoding is the representation of digital information by a digital
signal. (eg. computer-to-printer)
01011101 Digital/digital
encoding
Unipolar
Digital/ digital encoding Polar
Bipolar
Unipolar
Digital transmission systems work by sending voltage pulses along a media link,
usually a wire or a cable. In most types of encoding, one voltage level stands for
binary 0 and another level stands for binary 1. The polarity of a pulse refers to whether
it is positive or negative.
Unipolar encoding is so named because it uses only one polarity. Therefore, only one
of the two binary states is encoded, usually the 1. The other state, usually 0, is
represented by zero voltage, or an idle line.
Unipolar encoding uses only one level of value.
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Amplitude
0 1 0 0 1 1 1 0
Time
1’s encoded as positive, 0’s are idle. Unipolar encoding is straight forward and
inexpensive to implement. However, it has two problems that make it unusable: DC
component and synchronisation.
DC component
Average amplitude is nonzero  creates a direct current (DC) component, when a
signal contains a DC component it cannot travel through media that cannot handle DC
components: e.g. microwaves or transformers.
Synchronisation
When a signal is unvarying, the receiver cannot determine the beginning and ending of
each bit. Therefore, Synchronisation problem in unipolar encoding can occur
whenever the data stream includes a long uninterrupted series of 1’s or 0’s.
Polar Encoding
Polar encoding uses two voltage levels: one positive and one negative. In most polar
encoding methods the average voltage level on the line is reduced and the DC
component problem of unipolar encoding is alleviated.
Polar
NRZ RZ Biphase
NRZ-L NRZ-I Manchester Differential

Manchester
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Non-Return-to-Zero (NRZ) Encoding
In NRZ encoding, the level of the signal is always either positive or negative. In NRZ
system if the line is idle it means no transmission is occurring at all.
 NRZ-L (Non-return-to-zero, Level)

In NRZ-L the level of the signal is dependant upon the state of the bit.
A positive voltage usually means the bit is 0, and negative voltage means the bit is a 1
(and vice versa).
 NRZ-I (Non-return-to-zero, Invert)

In NRZ-I, an inversion of the voltage level represents a 1 bit. It is the transition
between a positive and a negative voltage, not the voltages themselves that represents
a 1 bit. A 0 bit is represented by no change.
An advantage of NRZ-I over NRZ-L is that because the signal changes every time a 1
bit is encountered, it provides some synchronisation.
Each inversion allows the receiver to synchronise its timer to the actual arrival of the
transmission.
Amplitude
0 1 0 0 1 1 1 0
NRZ-L
Time
NRZ-I Time
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RZ (Return-to-zero) Encoding
To assure synchronisation, there must be a signal change for each bit. The receiver can
use these changes to built up, update, and synchronise its clock.
One solution is return to zero (RZ) encoding, which uses three Values: positive,
negative, and zero.
Amplitude
0 1 0 0 1 1 1 0
Time
The main disadvantage of RZ encoding is that it requires two signal changes to encode
one bit and therefore occupies more bandwidth. But of the three alternatives discussed
above, it is the most effective. Because a good encoded digital signal must contain a
provision for synchronisation.
Biphase Encoding
Probably the best existing solution to the problem of synchronisation is biphase
encoding. In this method, the signal changes at the middle of the bit interval but does
not return to zero. Instead it continues to the opposite pole. As in RZ, these mid-
interval transitions allow for synchronisation.
Biphase encoding is implemented in two different ways: Manchester and differential
Manchester.
 Manchester
Manchester encoding uses the inversion at the middle of each bit interval for both
synchronisation and bit representation. A negative-to-positive transition represents
binary 1 and a positive-to-negative transition represents binary 0.
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Amplitude 0 1
0 1 0 0 1 1 1 0
Time
 Differential Manchester
In this method, the inversion at the middle of the bit is used for synchronisation, but
the presence or absence of an additional transition at the beginning of the interval is
used to identify the bit. A transition means binary 0 and no transition means binary 1.
The bit representation is shown by the inversion and non-inversion at the beginning of
the bit.
Amplitude
0 1 0 0 1 1 1 0
Time
Bipolar Encoding
Bipolar encoding uses three voltage levels: positive, negative and zero. The zero level
is used to represent binary 0 positive and negative voltages represent alternating 1s. (If
1st one +ve, 2nd is -ve).
* Three types of bipolar encoding are popular use by the data communications
industry: AMI, B8ZS, and HDB3.
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 Bipolar Alternate Mark Inversion (AMI)

Bipolar AMI is the simplest type of bipolar encoding. The word mark comes from
telegraphy and means 1.
AMI means alternate 1 inversion. A neutral, zero voltage represents binary 0. Binary
1s are represented by alternating positive and negative voltages
Amplitude
0 1 0 0 1 1 1 0
Time
By inverting on each occurrence of a 1, bipolar AMI accomplishes two things: first,

the DC component is zero, and second, a long sequence of 1s stays synchronised.
Two variations of bipolar AMI have been developed to solve the problem of
synchronisation sequential 0s. The first used in North America, is called bipolar 8-zero
substitution (B8ZS); the second, used in Europe and Japan, is called high-density
bipolar 3 (HDB3). Both are adaptations of bipolar AMI that modify the original
pattern only in the case of multiple consecutive 0s.
Bipolar 8-Zero Substitution (B8ZS)
B8ZS is the convention adopted in North America to provide synchronisation of long

strings of 0s. In most situations B8ZS functions identically to bipolar AMI. Bipolar
AMI changes poles with every 1 it encounters. These changes provide the
synchronisation needed by the receiver, but the signal does not change during a string
of 0s, so synchronisation is lost. The solution provided by B8ZS is to force artificial
signal changes, called violations
 In B8ZS, if eight 0s come one after another, we change the pattern in one of two ways
based on the polarity of previous 1.
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Polarity of Polarity of previous 1 bit

Previous 1 bit
+ 0 0 0 0 0 0 0 0 - 0 0 0 0 0 0 0 0
will change to
+ 0 0 0 + - 0 - + - 0 0 0 - + 0 + -
(Violation) (Violation) (Violation) (Violation)
High-Density Bipolar 3 (HDB3)
In HDB3 if four 0s come one after another, we change the pattern in one of four ways
based on the polarity of the previous 1 and the number of 1s since the last substitution.
+ 0 0 0 0 - 0 0 0 0
Violation in the 4th consecutive zero
+ 0 0 0 + - 0 0 0 -
If the number of 1s since the last substitution is odd
+ 0 0 0 0 - 0 0 0 0
Violation in the 1st & 4th consecutive zero
+ - 0 0 - - + 0 0 +
If the number of 1s since the last substitution is even
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Ex
Compare the bandwidth needed for unipolar encoding and RZ encoding. Assume the
worst-case scenario for both.
Solution
The worst case scenario (the situation requiring the most bandwidth) is alternating 1s
and 0s for unipolar, for RZ the worst-case is all 1s.
Unipolar encoding
Value
1 0 1 0 1 0 1 0
Time
Time
Value
RZ encoding
Time
Time
RZ needs twice the bandwidth of unipolar.
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Ex
Compare the bandwidth needed for Manchester and Differential Manchester encoding.
Assume the worst-case scenario for both.
Solution
The worst-case scenario for Manchester is consecutive 1s or consecutive 0s. There are
two transistors for each bit (one cycle per bit). For Differential Manchester the worst –
case is consecutive 0s with two transitions per each bit (one cycle per bit). The
bandwidths, which are proportional to bit rate, are the same for each.
Ex
Using B8ZS, encode the bit stream 10000000000100. Assume that the polarity of the
previous 1 is positive.
Amplitude
1 0 0 0 0 0 0 0 0 0 0 1 0 0
Time
Ex
Using HDB3, encode 10000000000100. Assume that the number of 1s so far is odd
and the previous 1 is positive.
Amplitude
1 0 0 0 0 0 0 0 0 0 0 1 0 0
Time
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Analog-to- Digital Encoding
Analog/digital
encoding
A / D encoder
(Coder-decoder)
In analog-to-digital encoding, the information contained in a continuous wave form are

represented as a series of digital pulses (1s and 0s).
Pulse Amplitude Modulation (PAM)

The first step in A/D encoding is called pulse amplitude modulation (PAM). This technique
takes analog information, samples it, and generates a series of pulses based on the results of
sampling. The term sampling means measuring the amplitude of the signal at equal time
intervals.
Amplitude Amplitude
Time Time
Analog Signal PAM Signal
In PAM, the original signal is sampled at equal intervals.

PAM has some applications, but it is not used by itself in data communications. However, it is
the first step in another very popular encoding method called pulse code modulation (PCM).
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Pulse Code Modulation (PCM)
PCM modifies the pulses created by PAM to create a complete digital signal. To do so, PCM
first quantises the PAM pulses. Quantisation is a method of assigning integral values in a
specific range to sampled instances. (The result of quantisation is presented in the following
figure).
+125
+100
+75
+50
+25
000
-25
-50
-75
-100
-125
Each value is translated into its seven-bit binary equivalent. The eighth bit indicates the sign.
+24 00011000 -15 10001111 +125 01111101
+38 00100110 -80 11010000 +110 01101110
+48 00110000 -50 10110010 +90 01011010
+39 00100111 +52 00110110 +88 01011000
+26 00011010 +127 01111111 +77 01001101
The binary digits are then transformed into a digital signal using one of the digital encoding.
PCM
0 0 0 1 1 0 0 0 0 0 1 0 0 1 1 0 0 0 1 1 0 0 0 0 …
Direction of transfer
The result of the PCM of the original signal encoded finally into a unipolar signal.
PCM is actually made up of four separate processes: PAM, quantisation, binary encoding, and
digital-to-digital encoding.
PCM is the sampling method used to digitize voice in T-line transmission in the North
America telecommunication system.
According to the Nyquist theorem, the sampling rate must be at least two times the highest
frequency.
Highest frequency = x Hz
Sampling rate = 2x samples/second

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Sampling interval
Quantisation
Quantising is the process of rounding-off the values of the flat-top samples to certain
predetermined levels.
u(t)
8
7
Quantising
6
5
4
3
2
1
T 2T 3T 4T 5T 6T t
Sampling
0111 0111 0100 0110 0110 0101 Binary Coding
PCM
Ex What sampling rate is needed for a signal with a bandwidth of 10,000 Hz (1000 Hz to
11,000 Hz)? If the quantisation is eight bits per sample, what is the bit rate?
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Solution
Sampling rate = 2 (11,000) = 22,000 samples/s each sample is quantised to eight bits: data
rate = (22,000 samples/s) (8 bits/sample) = 176 kbps
Digital-to-Analog Encoding
01011101 Digital/analog
encoding
Digital-to-analog encoding is the representation if digital information by an analog signal.
Digital /analog
encoding
ASK FSK PSK
QAM
Quadrature Amplitude Modulation
Bit Rate and Baud Rate

- Bit rate is the number of bits transmitted in one second.
- Baud rate refers to the number of signal units per second that are required to represent those
bits.
- For computer efficiency, bit rate is more important.
- For data transmission, baud rate is more important the fewer the signal units required, the
more efficient the system, and the less bandwidth required to transmit more bits.
Carrier signal
In analog transmission the sending device produces high - frequency signal that acts as a basis
for the information signal. The base signal is called the carrier signal or carrier frequency. The
receiving device is tuned to the frequency of the carrier signal that it expects from the sender.
Digital information is then encoded onto the carrier signal by modifying one or more of its
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characteristic (amplitude, frequency or phase). This kind of modification is called modulation
(or shift keying) and the information signal is called a modulating signal.
Amplitude Shift Keying (ASK)

In ASK the strength of the signal is varied to represent binary 1 or 0. Both frequency and
phase remain constant, while the amplitude changes.
Bit rate: 5 Baud rate: 5

Amplitude
1 bit 1 bit 1 bit 1 bit 1 bit
0 1 0 1 0
ASK Time
1 baud 1 baud 1 baud 1 baud 1 baud
1 second
0 1 0 1 1 0 0 1
b(t)
c(t) t
ASK t
Tb
1 second
Nbit = Nbaud = 8
Bit duration is the period of time that defines one bit. The peak amplitude of the signal, during
each bit duration, is constant and its value depends on the bit (0 or 1). The transmission speed
using ASK is limited by the physical characteristics of the transmission medium.
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Bandwidth for ASK
BW= (1 + d)*Nbaud
Where
BW is the bandwidth
Nbaud is the baud rate
d is a factor related to the condition of the line (with min. value of 0)
- The minimum bandwidth required for transmission is equal to the baud rate.
Amplitude
minimum bandwidth = Nbaud
Frequency
(fc-Nbaud/2) fc (fc+Nbaud/2)
Ex Find the bandwidth for an ASK signal transmitting at 2000 bps. Transmission is in half-
duplex mode.
Solution
In ASK baud rate = bit rate
Nbaud = 2,000
An ASK signal requires a bandwidth equal to its baud rate:
BW = 2,000 Hz.
Ex Given a bandwidth of 10,000 Hz (1,000 to 11,000 Hz), draw the full-duplex ASK diagram
of the system. Find the carriers and the bandwidth in each direction. Assume there is no gap
between the bands in two directions.
Solution
For full-duplex ASK the bandwidth for each direction is BW=10,000/2=5000 Hz.
The carrier frequencies can be chosen at the middle of each band
ƒc (forward) = 1,000 + 5,000/2 = 3,500 Hz
ƒc (backward) = 11,000-5,000/2 = 8,500 Hz
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Amplitude
ƒc (forward) ƒc (backward)
Frequency
1,000 3,500 6,000 8,500 11,000
Frequency Shift Keying (FSK)

In frequency shift keying (FSK), the frequency of the signal is varied to represent binary 1 or
0. The frequency of the signal during each bit duration is constant and its value depends on
the bit (0 or 1): both peak amplitude and phase remain constant.

Amplitude
0 1 0 1 0
FSK Time
1 second
0 1 0 1 1 0 0 1
b(t)
c1(t) t
c2(t) t
FSK t
Tb
1 second
Nbit = Nbaud = 8
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FSK avoids most of the noise problems of ASK. The limiting factors of FSK are the physical
capabilities of the carrier.
Bandwidth for FSK
FSK spectrum can be considered as the combinations of two ASK spectra centred on ƒc0 and
ƒc1. The bandwidth required for FSK transmission is equal to the baud rate of the signal plus
the frequency shift (difference between the two carrier frequencies).
Amplitude
BW = Nbaud + (ƒc1 - ƒc0)
fc1 - fc0
Frequency
fc0 fc1
Ex Find the bandwidth for an FSK signal transmitting at 2,000 bps. Transmission is in half-
duplex mode and the carriers must be separated by 3,000 Hz.
Solution
= 2,000 + 3,000 = 5,000 Hz.
Ex Find the maximum bit rate for an FSK signal if the bandwidth of the medium is
12,000 Hz and the distance between the two carriers must be at least 2,000 Hz. Transmission
is in full-duplex mode.
Solution
Because the transmission is in full-duplex, only 6,000 Hz is allocated for each direction, for
FSK, if ƒc1 and ƒc0 are the carrier frequencies,
Nbaud = BW – (ƒc1 - ƒc0)
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= 6,000 – 2,000 = 4,000
But because the baud rate is the same is bit rate, the bit rate is 4,000 bps.
Phase Shift Keying (PSK)
In the PSK, the phase is varied to represent binary 1 or 0. Both peak amplitude and frequency
remain constant as the phase changes. The phase of the signal during each bit duration, is
constant and its value depends on the bit (0 or 1).

Amplitude

0 1 0 1 0
PSK Time
1 second
0 1 0 1 1 0 0 1
b(t)
c(t) t
PSK t
DPSK t
1 second
Nbit = Nbaud = 8
DPSK eliminates the need for a coherent reference signal at the receiver by combing two
basic operations at the transmitter:
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- Differential encoding of the input data
- PSK DPSK
- To send symbol 1 we phase advance the current signal waveform by 180°.
- To send symbol 0 we leave the phase of the current signal waveform unchanged.
PSK Constellation
Bit Phase
0 0o 1 0
1 180°
Constellation diagram
The above method is often called 2-PSK, or binary PSK, because two different phases (0° and
180°) are used in the encoding.
PSK is not susceptible (easily influenced) to the noise degradation that affects ASK, nor to the
bandwidth limitations of FSK. This means that smaller variations in the signal can be detected
reliably by the receiver. Therefore instead of utilising only two variations of a signal, each
representing one bit, we can use four variations and let each phase shift represent two bits.
Bit rate: 8 bps Baud rate: 4

Amplitude
2 bits 2 bits 2 bits 2 bits

00 01 10 11
Time
0o 90o 180o 270o

1 baud
1 second
4-PSK (Quadrature-PSK)
This technique is called 4-PSK or Q-PSK. The pair of bits represented by each phase is called
a dibit.
Data can be transmitted two times as fast using 4-PSK as using 2-PSK.
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4–PSK characteristics 01
Dibit Phase
00 0o
01 90o 10 00
10 180o
11 270o
11
8–PSK Characteristics Constellation diagram
Tribit Phase 010
000 0o 011 001
001 45o
010 90o
011 135o 100 000
100 180o
101 225o
110 270o 101 111
111 315o
110
Bit rate of 8-PSk is three as that of 2-PSK Constellation diagram
Bandwidth for PSK

The min. BW required for PSK transmission is the same as that required for ASK
transmission.
BW = Nbaud
Ex: Find the bandwidth for a 4-PSK signal transmitting at 2,000 bps. Transmission is in half-
duplex mode.
Solution
For 4-PSK the baud rate is half of the bit rate.
The baud rate is therefore 1,000. A PSK signal requires a bandwidth equal to its baud rate.
Therefore the bandwidth is 1,000 Hz.
Ex: Given a bandwidth of 5,000 Hz for an 8-PSK signal, what are the baud rate and bit rate?
Solution
For PSK the baud rate is the same as the bandwidth, which means the baud rate is 5,000. But
in 8-PSK, the bit rate is three times the baud rate. So the bit rate is 15,000 bps.
Quadrature Amplitude Modulation (QAM)

QAM means combining ASK and PSK in such a way that we have maximum contrast
between each bit, dibit, tribit, quadbit, and so on.
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Possible variation of QAM is numerous; theoretically any measurable number of changes in
amplitude can be combined with any measurable number of changes in phase.
The figure below shows the constellation diagrams of 4-QAM and 8-QAM. In both cases the
number of amplitude shifts is less than the number of phase shifts. Because amplitude
changes are susceptible to noise and require greater shift differences than do phase changes,
the number of phase shifts used by a QAM system is always larger than the number of
amplitude shifts.
011
01 00 010
101 100 000 001
110
10 11
111
4-QAM 8-QAM
Constellation Diagram Constellation Diagram
Amplitude
Nbit = 24 bps, Nbaud = 8
3 bits 3 bits 3 bit 3 bits 3 bits 3 bits 3 bits 3 bits
101 100 001 000 010 011 110 111
1 baud
The output signal of the 8-QAM modem for data

b (t) = 101 100 001 000 010 011 110 111
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Three popular 16-QAM configurations are shown bellow:
4 amplitudes, 8 phase 3 amplitudes, 12 phases 2 amplitude, 8 phase
16-QAM 16-QAM 16-QAM
Since amplitude shift is more susceptible to noise, the greater the ratio of phase shifts to
amplitude, the greater the immunity to noise.
The second example, three amplitudes and 12 phases, handles noise best. The first
example, 4 amplitudes and 8 phases, is the OSI (Open Systems Interconnection)
recommendation. Several QAM designs link specific amplitudes with specific phases.
This means that even with noise problems associated with amplitude shifting, the meaning
of a shift can be recovered from phase information.
Bandwidth for QAM

The minimum bandwidth required for QAM transmission is the same as that required for
ASK and PSK transmission. QAM has the same advantages of PSK over ASK.
Bit/Baud Comparison
Assuming that an FSK signal over voice-grade phone lines can send 1200 bps, the bit rate
is 1200 bps. Each frequency shift represents a single bit; so it requires 1200 signal
elements to send 1200 bits. Its baud rate, therefore, is also 1200. Each signal variation in
8-QAM system, however, represents three bits. So a bit rate of 1200 bps, using 8-QAM,
has a baud rate of only 400.
As the figure below shows, a dibit system has a baud rate of one-half the bit rate, a tribit
system has a baud rate of one-third the bit rate, a quadbit system has a baud rate of one-
fourth the bit rate.
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Bit
Nbaud= N Nbit= N
0 0 1 0 1 0 0 0 1 0 1 0 1 1 1 0
Dibit
Nbaud= N Nbit= 2N
0 0 1 0 1 0 0 0 1 0 1 0 1 1 1 0
Tribit
Nbaud= N Nbit= 3N
0 0 1 0 1 0 0 0 1 0 1 0 1 1 1 0
Quadbit
Nbaud= N Nbit= 4N
0 0 1 0 1 0 0 0 1 0 1 0 1 1 1 0
Bit/ Baud Rate Comparison

Modulation Units Bits/Bauds Baud Rate Bit Rate
ASK, FSK, 2-PSK Bit 1 N N
4-PSK, 4-QAM Dibit 2 N 2N
8-PSK, 8-QAM Tribit 3 N 3N
16-QAM Quadbit 4 N 4N
32-QAM Pentabit 5 N 5N
64-QAM Hexabit 6 N 6N
128-QAM Septabit 7 N 7N
256-QAM Octabit 8 N 8N
Ex
A constellation diagram consists of eight equally spaced points on a circle. If the bit rate is
4800 bps, what is the baud rate?
Solution
The constellation indicates 8-PSK with points 45o apart. Since 23 = 8, three bits are
transmitted with each signal element. Therefore, the baud rate is
4800/3 = 1600 baud
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Ex
Compute the baud rate for a 72,000 bps 64-QAM.
Solution
A 64-QAM signal means that there are six bits per signal elements since 26 = 64. Thus,
72,000/6 = 12,000 baud
Ex
Compute the bit rate for a 1,000 baud 16-QAM signal.
Solution
A 16-QAM signal means that there are four bits per signal elements since 24 = 16. Thus,
(1,000)(4) = 4,000 bps
Analog-to-Analog-Encoding
Analogl/analog
encoding
Analog-to-analog encoding is the representation of analog information by an analog signal.

(eg. Radio communication).
Analog-to-analog modulation can be accomplished in three ways: amplitude modulation
(AM), frequency modulation (FM), and phase modulation (PM)
Analogl/analog
encoding
AM FM PM
Amplitude Modulation
In AM transmission, the carrier signal is modulated so that its amplitude varies with the
changing amplitudes of the modulating signal. The frequency and phase of the carrier remain
the same; only the amplitude changes to follow variations in the information. The modulating
signal becomes the envelope of the carrier.
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Amplitude
Modulating signal (audio)
Time
Carrier frequency
Amplitude
Time
AM signal
Amplitude
Time
AM Bandwidth
The bandwidth of an AM signal is equal to twice the bandwidth of the modulating signal and
covers a range centred on the carrier frequency. The total bandwidth required for AM can be
determined from the bandwidth of the audio signal:
BWt = 2 x BWm
Amplitude
ƒc Frequency
BWm BWm
BWt = 2 x BWm
BWm = Bandwidth of the modulating signal (audio)
BWt = Total bandwidth (radio)
ƒc = Frequency of the carrier
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The bandwidth of an audio signal (speech & music) is usually 5 kHz. Therefore, an AM radio
station needs a minimum bandwidth of 10 kHz. In fact, the Federal Communications
Commission (FCC) allows 10 kHz for each AM station.
AM stations are allowed carrier frequencies anywhere between 530 and 1700 kHz (1.7 MHz).
However, each station’s carrier frequency must be separated from those on either side by at
least 10 kHz (one AM bandwidth) to avoid interference.
fc fc fc fc fc
No station here
No station here
No station here
530 1700
kHz 10 kHz
kHz
Frequency Modulation (FM)
In FM transmission, the frequency of the carrier signal is modulated to follow the changing
voltage level (amplitude) of the modulating signal. The peak amplitude and phase of the
carrier signal remain constant, but as the amplitude of the information signal changes, the
frequency of the carrier changes correspondingly.
Amplitude Modulating signal (audio)
Time
Carrier frequency
Amplitude
Time
Amplitude
FM signal
Time
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FM Bandwidth
The bandwidth of an FM signal is equal to 10 times the bandwidth of the modulating signal
and, like AM bandwidth, covers a range centred on the carrier frequency. The total bandwidth
required for FM can be determined from the bandwidth of the audio signal:
BWt = 10 x BWm
Amplitude
ƒc Frequency
5BWm 5BWm
BWt = 10 x BWm
BWm = Bandwidth of the modulating signal (audio)

BWt = Total bandwidth (radio)
ƒc = Frequency of the carrier
The bandwidth of an audio signal (speech & music) broadcast in stereo is almost 15 kHz.
Therefore, each FM radio station needs a minimum bandwidth of 150 kHz. The FCC allows
200 kHz (0.2 MHz) for each FM station to provide some room for guard bands.
FM stations are allowed carrier frequencies anywhere between 88 and 108 MHz. However,
stations must be separated from by at least 200 kHz to avoid overlapping.
fc fc fc fc fc
No station here
No station here
No station here
88 MHz 108 MHz

200 kHz
Phase Modulation (PM)
Due to simpler hardware requirements, PM is used in some systems as an alternative to FM.
In PM transmission, the phase of the carrier signal is modulated to follow the changing
voltage level of the modulating signal. The peak amplitude and frequency of the carrier signal
remain constant, but as the amplitude of the information signal changes, the phase of carrier
changes correspondingly. The analysis and final result (modulating signal) are similar to those
of FM.
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TRANSMISSION CODES
Transmission Codes
Binary-Coded Decimal (also called 8421 BCD)
In BCD, four bits are used to encode one decimal character. Four bits give 16 binary
combinations. Since there are 10 decimal characters, 0 through 9, only 10 of the 16 possible
combinations are necessary for encoding in BCD. The remaining 6 combinations are said to
be invalid.
Decimal BCD
0 0000
1 0001
2 0010 1010
3 0011 1011
4 0100 1100
1101 Not valid in BCD
5 0101
6 0110 1110
7 0111 1111
8 1000
9 1001
Ex
Convert 36710 to BCD
Solution
36710 = 0011 0110 0111
Ex
Convert 124910 to BCD
Solution
124910 = 0001 0010 0100 1001
Ex
Convert 5810 to BCD
Solution
5810 = 0101 1000
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BCD Addition:
Straight binary addition is performed as long as the result does not exceed a decimal value
of 9.
Ex
Add the decimal numbers 3 and 4 in BCD
Solution
3 0011
+4 0100
7 0111
Ex
Solution
63 0110 0011
+ 24 0010 0100
87 1000 0111
When the sum of two numbers exceeds 9, an invalid BCD number is obtained. The invalid
number can be converted to a valid number by adding 0110 (6) to it.
Ex
Solution
9 1001
+6 0110
15 1111…not valid in BCD

+ 0110…add 6 for correction
0001 0101…correct BCD number
Ex
Solution
46 0100 0110
+ 79 0111 1001
125 1011 1111

+ 0110 0110
0001 0010 0101
1 2 5
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Excess-3 Code
Excess-3 code is very similar to 8421 BCD code. The only difference is that 3 is added to the
decimal before it is encoded into a four-word.
Decimal BCD Excess-3

0 0000 0011
1 0001 0100
2 0010 0101 0000
3 0011 0110 0001
4 0100 0111 0010
5 0101 1000 1101 Not valid in Excess-3
6 0110 1001 1110

7 0111 1010 1111
8 1000 1011
9 1001 1100
Ex
Add the decimal numbers 9 and 7 in Excess-3
Solution
9 1100
+7 1010
16 1 0110
Gray Code
The disadvantage of the previous codes is that several bits change state between adjacent
counts. The Gray code is unique in that successive counts result in only one bit change. For
example, 7 (0111) to 8 (1000) in binary, or BCD, all four bits change state. In Gray code,
however, 7 (0100) to 8 (1100) require a single bit change.
The switching noise generated by the associated circuits may be intolerable in some
environments. The same change with Gray code undergoes only a single bit change
consequently, less noise is generated. Shaft encoders used for receiver tuning dials often use
Gray code.
The Gray code is widely used for encoding the position of the rotary shaft and for data
transmission using PSK.
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Decimal Binary Gray Code
0 0000 0000
1 0001 0001
2 0010 0011
3 0011 0010
4 0100 0110
5 0101 0111
6 0110 0101
7 0111 0100
8 1000 1100
9 1001 1101
10 1010 1111
Binary-to-Gray Conversion
- The given binary code is shifted to the right by one bit.
- Discard the last bit (the LSB) from the obtained bits.
- Exclusive-ORing the given and obtained bits result in the equivalent Gray code.
MSB LSB
x x x - - - - - x x
16th position 1st position

Ex
Compute the Gray code for the binary number 11010
Solution
Binary code 1 1 0 1 0
 1 1 0 1
Gray code 1 0 1 1 1
Ex
Compute the Gray code for the binary number 10001101
Solution
Binary code 1 0 0 0 1 1 0 1
 1 0 0 0 1 1 0
Gray code 1 1 0 0 1 0 1 1
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Gray-to-Binary Conversion
- The first bit, the leftmost of the given Gray code, becomes the MSB of the Binary
code.
- Exclusive-ORing the second Gray code bit with the MSB of the binary code yields the
second binary bit.
- Exclusive-ORing the third Gray code bit with the second binary code yields the third
binary bit.
- Exclusive-ORing the fourth Gray code bit with the third binary code yields the fourth
binary bit. And so on.
Ex
Compute the binary code for the Gray code 101101
Solution
Gray code 1 0 1 1 0 1
    
Binary code 1 1 0 1 1 0
Ex
Compute the binary code for the Gray code 11001011
Solution
Gray code 1 1 0 0 1 0 1 1
      
Binary code 1 0 0 0 1 1 0 1
Binary Numbers
The binary numbering system provides the basis for all computer operations. Computers
work by manipulating electrical current on and off. The binary system uses two symbols,
0 and 1. Also called base 2.
Binary weights
Position Fifth Fourth Third Second First
Weight 24 (16) 23 (8) 22 (4) 21 (2) 20 (1)
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Ex
1 1 0 1 digits
8 4 2 1 weights
___________________
8 4 0 1 results
13
Octal Numbers
The octal numbering system is used by computer programmers to represent binary
numbers in compact form. Also called base 8.
Octal numbers use 8 symbols: 0,1,2,3,4,5,6,7.
Octal weights
Weight 84 (4096) 83 (512) 82 (64) 81 (8) 80 (1)
Ex
3 4 7 1 digits
512 64 8 1 weights
___________________
1,536 256 56 1 results
1,849
Hexadecimal Numbers
Hexadecimal numbering system, like octal, is used by computer programmers to represent
binary numbers in compact form. Also called base 16.
Hexadecimal uses 16 symbols: 0,1,2,3,4,5,6,7,8,9,A,B,C,D,E,F.
Hexadecimal weights
Weight 164 (65,536) 163 (4,096) 162 (256) 161 (16) 160 (1)
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Ex
3 4 7 1 digits
4,096 256 16 1 weights
_____________________
12,288 1,024 112 1 results
+
13,425
Decimal Binary Octal Hexadecimal
0 0000 0 0
1 0001 1 1
2 0010 2 2
3 0011 3 3
4 0100 4 4
5 0101 5 5
6 0110 6 6
7 0111 7 7
8 1000 10 8
9 1001 11 9
10 1010 12 A
11 1011 13 B
12 1100 14 C
13 1101 15 D
14 1110 16 E
15 1111 17 F
Transformations
- From Other Systems to Decimal
a) From Binary to Decimal
1 0 0 1 1 1 0 Binary
64 32 16 8 4 2 1 weights
64 0 0 8 4 2 0 weighted results
78
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b) Hexadecimal to Decimal
4 E Hexadecimal
16 1 weights
64 14 weighted results
78
c) Octal to Decimal
1 1 6 Octal
64 8 1 weights
64 8 6 weighted results
78
- From Binary to Octal or Hexadecimal
Group by three Group by four

1 0 0 1 1 1 0 From right to from right to left 1 0 0 1 1 1 0
left
1 1 6 4 E
Octal Hexadecimal
- From Octal or Hexadecimal to Binary
Octal Hexadecimal
1 1 6 4 E
1 0 0 1 1 1 0 1 0 0 1 1 1 0
Binary Binary
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- From Decimal to Other Systems
a) From Decimal to Binary (divide by 2) Decimal
Remainder
0 1 2 4 9 19 39 78
1 0 0 1 1 1 0 Binary
Code
b) From Decimal to Octal (divide by 8)
Remainder
0 1 9 78
1 1 6 Binary
Code
c) From Decimal to Hexadecimal (divide by 16)
Remainder
0 4 78
4 E Binary
Code
Morse Code
Morse code is one of the oldest electrical transmission codes. The digital code system is made
up of a series of dots and dashes, representing the alphabet and decimal numbers system. A
dash is three times the duration of a dot.
A
B
C
D
.
.
1
2
.
ASCII Code
The American Standard Code for Information Interchange (ASCII) is the most widely used
alphanumeric code for transmission and data processing.
ASCII is a seven-bit code that can be represented by two hexadecimal characters for
simplicity. The MS hexadecimal character in this case never exceed 7
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Ex
What is the ASCII code for the letter H (uppercase) in binary and hexadecimal?
Solution
Letter H is located in column 4 and row 8.
Binary: 100 1000
Hexadecimal: $ 48
Ex
What is the ASCII code for the letter k (lowercase) in binary and hexadecimal?
Solution
Letter k is located in column 6 and row B:
Binary: 110 1011
Hexadecimal: $ 6B
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UNIT-5
ADVANCED MOBILE PHONE SYSTEM
 Advanced Mobile Phone Service (AMPS) is a standard system

for analog signal cellular telephone service in the United States
and is also used in other countries.
 It is based on the initial electromagnetic radiation spectrum

allocation for cellular service by the Federal Communications
Commission (FCC) in 1970. Introduced by AT&T in 1983,
AMPS became one of the most widely deployed cellular system
in the United States.
 AMPS allocates frequency ranges within the 800 and 900

Megahertz (MHz) spectrum to cellular telephone. Each service
provider can use half of the 824-849 MHz range for receiving
signals from cellular phones and half the 869-894 MHz range
for transmitting to cellular phones.
 The bands are divided into 30 kHz sub-bands, called channels.

The receiving channels are called reverse channels and the
sending channels are called forward channels. The division of
the spectrum into sub-band channels is achieved by using
frequency division multiple access (FDMA).
 The signals received from a transmitter cover an area called a

cell. As a user moves out of the cell's area into an adjacent cell,
the user begins to pick up the new cell's signals without any
noticeable transition.
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 The signals in the adjacent cell are sent and received on

different channels than the previous cell's signals to so that the
signals don't interfere with each other.
 The analog service of AMPS has been updated with digital

cellular service by adding to FDMA a further subdivision of
each channel using time division multiple access (TDMA). This
service is known as digital AMPS (D-AMPS). Although AMPS
and D-AMPS originated for the North American cellular
telephone market , they are now used worldwide with over 74
million subscribers, according to Ericsson, one of the major
cellular phone manufacturers.
GSM
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GSM MSs consist of:

• Mobile Equipment
• Subscriber Identity Module
FUNCTIONS OF MS
• Voice and data transmission & receipt
• Frequency and time synchronization
• Monitoring of power and signal quality of the surrounding cells
• Provision of location updates even during inactive state
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CDMA
Code division multiple access
From Wikipedia, the free encyclopedia

This article is about a channel access method. For the mobile phone
technology referred to as CDMA, see IS-95 and CDMA2000.
Multiplex techniques
Analog modulation
 AM
 FM
 PM
 QAM
 SM
 SSB
Circuit mode (constant bandwidth)
 TDM
 FDM / WDM
 SDM
 Polarization multiplexing
 Spatial multiplexing
 OAM multiplexing
Statistical multiplexing (variable

bandwidth)
 Packet switching
 Dynamic TDM
 FHSS
 DSSS
 OFDMA
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 SC-FDM
 MC-SS
Related topics
 Channel access methods

 Media access control
 v
 t
 e
Code division multiple access (CDMA) is a channel access method

used by various radio communication technologies.
CDMA is an example of multiple access, which is where several

transmitters can send information simultaneously over a single
communication channel. This allows several users to share a band of
frequencies (see bandwidth). To permit this to be achieved without
undue interference between the users, CDMA employs spread-
spectrum technology and a special coding scheme (where each
transmitter is assigned a code).
CDMA is used as the access method in many mobile phone standards

such as cdmaOne, CDMA2000 (the 3G evolution of cdmaOne), and
WCDMA (the 3G standard used by GSM carriers), which are often
referred to as simply CDMA.
ode division multiplexing (synchronous CDMA)
The digital modulation method is analogous to those used in simple

radio transceivers. In the analogue case, a low frequency data signal is
time multiplied with a high frequency pure sine wave carrier, and
transmitted. This is effectively a frequency convolution (Weiner-
Kinchin Theorem) of the two signals, resulting in a carrier with
narrow sidebands. In the digital case, the sinusoidal carrier is replaced
by Walsh functions. These are binary square waves that form a
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complete orthonormal set. The data signal is also binary and the time
multiplication is achieved with a simple XOR function. This is
usually a Gilbert cell mixer in the circuitry.
Synchronous CDMA exploits mathematical properties of

orthogonality between vectors representing the data strings. For
example, binary string 1011 is represented by the vector (1, 0, 1, 1).
Vectors can be multiplied by taking their dot product, by summing the
products of their respective components (for example, if u = (a, b) and
v = (c, d), then their dot product u·v = ac + bd). If the dot product is
zero, the two vectors are said to be orthogonal to each other. Some
properties of the dot product aid understanding of how W-CDMA
works. If vectors a and b are orthogonal, then and:
Each user in synchronous CDMA uses a code orthogonal to the

others' codes to modulate their signal. An example of four mutually
orthogonal digital signals is shown in the figure. Orthogonal codes
have a cross-correlation equal to zero; in other words, they do not
interfere with each other. In the case of IS-95 64 bit Walsh codes are
used to encode the signal to separate different users. Since each of the
64 Walsh codes are orthogonal to one another, the signals are
channelized into 64 orthogonal signals. The following example
demonstrates how each user's signal can be encoded and decoded.
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Example
An example of four mutually orthogonal digital signals.
Start with a set of vectors that are mutually orthogonal. (Although

mutual orthogonality is the only condition, these vectors are usually
constructed for ease of decoding, for example columns or rows from
Walsh matrices.) An example of orthogonal functions is shown in the
picture on the right. These vectors will be assigned to individual users
and are called the code, chip code, or chipping code. In the interest of
brevity, the rest of this example uses codes, v, with only two bits.
Each user is associated with a different code, say v. A 1 bit is

represented by transmitting a positive code, v, and a 0 bit is
represented by a negative code, –v. For example, if v = (v0, v1) = (1, –
1) and the data that the user wishes to transmit is (1, 0, 1, 1), then the
transmitted symbols would be
(v, –v, v, v) = (v0, v1, –v0, –v1, v0, v1, v0, v1) = (1, –1, –1, 1, 1, –1, 1, –
1). For the purposes of this article, we call this constructed vector the
transmitted vector.
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Each sender has a different, unique vector v chosen from that set, but
the construction method of the transmitted vector is identical.
Now, due to physical properties of interference, if two signals at a

point are in phase, they add to give twice the amplitude of each signal,
but if they are out of phase, they subtract and give a signal that is the
difference of the amplitudes. Digitally, this behaviour can be
modelled by the addition of the transmission vectors, component by
component.
If sender0 has code (1, –1) and data (1, 0, 1, 1), and sender1 has code
(1, 1) and data (0, 0, 1, 1), and both senders transmit simultaneously,
then this table describes the coding steps:
Step Encode sender0 Encode sender1
0 code0 = (1, –1), data0 = (1, 0, code1 = (1, 1), data1 = (0, 0, 1,
1, 1) 1)
1 encode0 = 2(1, 0, 1, 1) – (1, 1, encode1 = 2(0, 0, 1, 1) – (1, 1, 1,

1, 1) = (1, –1, 1, 1) 1) = (–1, –1, 1, 1)
2 signal0 = encode0 ⊗ code0 signal1 = encode1 ⊗ code1

= (1, –1, 1, 1) ⊗ (1, –1) = (–1, –1, 1, 1) ⊗ (1, 1)
= (1, –1, –1, 1, 1, –1, 1, –1) = (–1, –1, –1, –1, 1, 1, 1, 1)
Because signal0 and signal1 are transmitted at the same time into the
air, they add to produce the raw signal:
(1, –1, –1, 1, 1, –1, 1, –1) + (–1, –1, –1, –1, 1, 1, 1, 1) = (0, –2, –
2, 0, 2, 0, 2, 0)
This raw signal is called an interference pattern. The receiver then

extracts an intelligible signal for any known sender by combining the
sender's code with the interference pattern, the receiver combines it
with the codes of the senders. The following table explains how this
works and shows that the signals do not interfere with one another:
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Step Decode sender0 Decode sender1
code0 = (1, –1), signal = (0, –2, code1 = (1, 1), signal = (0, –2,
0
–2, 0, 2, 0, 2, 0) –2, 0, 2, 0, 2, 0)
1 decode0 = pattern.vector0 decode1 = pattern.vector1
decode0 = ((0, –2), (–2, 0), (2, decode1 = ((0, –2), (–2, 0), (2,
2
0), (2, 0)).(1, –1) 0), (2, 0)).(1, 1)
decode0 = ((0 + 2), (–2 + 0), (2 decode1 = ((0 – 2), (–2 + 0), (2
3
+ 0), (2 + 0)) + 0), (2 + 0))
data0=(2, –2, 2, 2), meaning (1, data1=(–2, –2, 2, 2), meaning

4
0, 1, 1) (0, 0, 1, 1)
Further, after decoding, all values greater than 0 are interpreted as 1

while all values less than zero are interpreted as 0. For example, after
decoding, data0 is (2, –2, 2, 2), but the receiver interprets this as (1, 0,
1, 1). Values of exactly 0 means that the sender did not transmit any
data, as in the following example:
Assume signal0 = (1, –1, –1, 1, 1, –1, 1, –1) is transmitted alone. The
following table shows the decode at the receiver:
Step Decode sender0 Decode sender1
code0 = (1, –1), signal = (1, –1, code1 = (1, 1), signal = (1, –1,
0
–1, 1, 1, –1, 1, –1) –1, 1, 1, –1, 1, –1)
1 decode0 = pattern.vector0 decode1 = pattern.vector1
decode0 = ((1, –1), (–1, 1), (1, – decode1 = ((1, –1), (–1, 1), (1, –
2
1), (1, –1)).(1, –1) 1), (1, –1)).(1, 1)
3 decode0 = ((1 + 1), (–1 – 1),(1 + decode1 = ((1 – 1), (–1 + 1),(1
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1), (1 + 1)) – 1), (1 – 1))
data0 = (2, –2, 2, 2), meaning data1 = (0, 0, 0, 0), meaning no

4
(1, 0, 1, 1) data
When the receiver attempts to decode the signal using sender1's code,
the data is all zeros, therefore the cross correlation is equal to zero and
it is clear that sender1 did not transmit any data.
Asynchronous CDMA
See also: Direct-sequence spread spectrum and near-far problem
When mobile-to-base links cannot be precisely coordinated,

particularly due to the mobility of the handsets, a different approach is
required. Since it is not mathematically possible to create signature
sequences that are both orthogonal for arbitrarily random starting
points and which make full use of the code space, unique "pseudo-
random" or "pseudo-noise" (PN) sequences are used in asynchronous
CDMA systems. A PN code is a binary sequence that appears random
but can be reproduced in a deterministic manner by intended
receivers. These PN codes are used to encode and decode a user's
signal in Asynchronous CDMA in the same manner as the orthogonal
codes in synchronous CDMA (shown in the example above). These
PN sequences are statistically uncorrelated, and the sum of a large
number of PN sequences results in multiple access interference (MAI)
that is approximated by a Gaussian noise process (following the
central limit theorem in statistics). Gold codes are an example of a PN
suitable for this purpose, as there is low correlation between the
codes. If all of the users are received with the same power level, then
the variance (e.g., the noise power) of the MAI increases in direct
proportion to the number of users. In other words, unlike synchronous
CDMA, the signals of other users will appear as noise to the signal of
interest and interfere slightly with the desired signal in proportion to
number of users.
All forms of CDMA use spread spectrum process gain to allow

receivers to partially discriminate against unwanted signals. Signals
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encoded with the specified PN sequence (code) are received, while

signals with different codes (or the same code but a different timing
offset) appear as wideband noise reduced by the process gain.
Since each user generates MAI, controlling the signal strength is an

important issue with CDMA transmitters. A CDM (synchronous
CDMA), TDMA, or FDMA receiver can in theory completely reject
arbitrarily strong signals using different codes, time slots or frequency
channels due to the orthogonality of these systems. This is not true for
Asynchronous CDMA; rejection of unwanted signals is only partial.
If any or all of the unwanted signals are much stronger than the
desired signal, they will overwhelm it. This leads to a general
requirement in any asynchronous CDMA system to approximately
match the various signal power levels as seen at the receiver. In
CDMA cellular, the base station uses a fast closed-loop power control
scheme to tightly control each mobile's transmit power.
Advantages of asynchronous CDMA over other techniques

Efficient practical utilization of the fixed frequency spectrum
In theory CDMA, TDMA and FDMA have exactly the same spectral
efficiency but practically, each has its own challenges – power control
in the case of CDMA, timing in the case of TDMA, and frequency
generation/filtering in the case of FDMA.
TDMA systems must carefully synchronize the transmission times of

all the users to ensure that they are received in the correct time slot
and do not cause interference. Since this cannot be perfectly
controlled in a mobile environment, each time slot must have a guard-
time, which reduces the probability that users will interfere, but
decreases the spectral efficiency. Similarly, FDMA systems must use
a guard-band between adjacent channels, due to the unpredictable
doppler shift of the signal spectrum because of user mobility. The
guard-bands will reduce the probability that adjacent channels will
interfere, but decrease the utilization of the spectrum.
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Flexible allocation of resources
Asynchronous CDMA offers a key advantage in the flexible

allocation of resources i.e. allocation of a PN codes to active users. In
the case of CDM (synchronous CDMA), TDMA, and FDMA the
number of simultaneous orthogonal codes, time slots and frequency
slots respectively are fixed hence the capacity in terms of number of
simultaneous users is limited. There are a fixed number of orthogonal
codes, time slots or frequency bands that can be allocated for CDM,
TDMA, and FDMA systems, which remain underutilized due to the
bursty nature of telephony and packetized data transmissions. There is
no strict limit to the number of users that can be supported in an
asynchronous CDMA system, only a practical limit governed by the
desired bit error probability, since the SIR (Signal to Interference
Ratio) varies inversely with the number of users. In a bursty traffic
environment like mobile telephony, the advantage afforded by
asynchronous CDMA is that the performance (bit error rate) is
allowed to fluctuate randomly, with an average value determined by
the number of users times the percentage of utilization. Suppose there
are 2N users that only talk half of the time, then 2N users can be
accommodated with the same average bit error probability as N users
that talk all of the time. The key difference here is that the bit error
probability for N users talking all of the time is constant, whereas it is
a random quantity (with the same mean) for 2N users talking half of
the time.
In other words, asynchronous CDMA is ideally suited to a mobile

network where large numbers of transmitters each generate a
relatively small amount of traffic at irregular intervals. CDM
(synchronous CDMA), TDMA, and FDMA systems cannot recover
the underutilized resources inherent to bursty traffic due to the fixed
number of orthogonal codes, time slots or frequency channels that can
be assigned to individual transmitters. For instance, if there are N time
slots in a TDMA system and 2N users that talk half of the time, then
half of the time there will be more than N users needing to use more
than N time slots. Furthermore, it would require significant overhead
to continually allocate and deallocate the orthogonal code, time slot or
frequency channel resources. By comparison, asynchronous CDMA
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transmitters simply send when they have something to say, and go off
the air when they don't, keeping the same PN signature sequence as
long as they are connected to the system.
Spread-spectrum characteristics of CDMA
Most modulation schemes try to minimize the bandwidth of this

signal since bandwidth is a limited resource. However, spread
spectrum techniques use a transmission bandwidth that is several
orders of magnitude greater than the minimum required signal
bandwidth. One of the initial reasons for doing this was military
applications including guidance and communication systems. These
systems were designed using spread spectrum because of its security
and resistance to jamming. Asynchronous CDMA has some level of
privacy built in because the signal is spread using a pseudo-random
code; this code makes the spread spectrum signals appear random or
have noise-like properties. A receiver cannot demodulate this
transmission without knowledge of the pseudo-random sequence used
to encode the data. CDMA is also resistant to jamming. A jamming
signal only has a finite amount of power available to jam the signal.
The jammer can either spread its energy over the entire bandwidth of
the signal or jam only part of the entire signal.[9]
CDMA can also effectively reject narrow band interference. Since

narrow band interference affects only a small portion of the spread
spectrum signal, it can easily be removed through notch filtering
without much loss of information. Convolution encoding and
interleaving can be used to assist in recovering this lost data. CDMA
signals are also resistant to multipath fading. Since the spread
spectrum signal occupies a large bandwidth only a small portion of
this will undergo fading due to multipath at any given time. Like the
narrow band interference this will result in only a small loss of data
and can be overcome.
Another reason CDMA is resistant to multipath interference is

because the delayed versions of the transmitted pseudo-random codes
will have poor correlation with the original pseudo-random code, and
will thus appear as another user, which is ignored at the receiver. In
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other words, as long as the multipath channel induces at least one chip
of delay, the multipath signals will arrive at the receiver such that they
are shifted in time by at least one chip from the intended signal. The
correlation properties of the pseudo-random codes are such that this
slight delay causes the multipath to appear uncorrelated with the
intended signal, and it is thus ignored.
Some CDMA devices use a rake receiver, which exploits multipath

delay components to improve the performance of the system. A rake
receiver combines the information from several correlators, each one
tuned to a different path delay, producing a stronger version of the
signal than a simple receiver with a single correlation tuned to the
path delay of the strongest signal.[10]
Frequency reuse is the ability to reuse the same radio channel

frequency at other cell sites within a cellular system. In the FDMA
and TDMA systems frequency planning is an important consideration.
The frequencies used in different cells must be planned carefully to
ensure signals from different cells do not interfere with each other. In
a CDMA system, the same frequency can be used in every cell,
because channelization is done using the pseudo-random codes.
Reusing the same frequency in every cell eliminates the need for
frequency planning in a CDMA system; however, planning of the
different pseudo-random sequences must be done to ensure that the
received signal from one cell does not correlate with the signal from a
nearby cell.[11]
Since adjacent cells use the same frequencies, CDMA systems have
the ability to perform soft hand offs. Soft hand offs allow the mobile
telephone to communicate simultaneously with two or more cells. The
best signal quality is selected until the hand off is complete. This is
different from hard hand offs utilized in other cellular systems. In a
hard hand off situation, as the mobile telephone approaches a hand
off, signal strength may vary abruptly. In contrast, CDMA systems
use the soft hand off, which is undetectable and provides a more
reliable and higher quality signal.[11]
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Collaborative CDMA
In a recent study, a novel collaborative multi-user transmission and

detection scheme called Collaborative CDMA[12] has been
investigated for the uplink that exploits the differences between users’
fading channel signatures to increase the user capacity well beyond
the spreading length in multiple access interference (MAI) limited
environment. The authors show that it is possible to achieve this
increase at a low complexity and high bit error rate performance in
ﬂat fading channels, which is a major research challenge for
overloaded CDMA systems. In this approach, instead of using one
sequence per user as in conventional CDMA, the authors group a
small number of users to share the same spreading sequence and
enable group spreading and despreading operations. The new
collaborative multi-user receiver consists of two stages: group multi-
user detection (MUD) stage to suppress the MAI between the groups
and a low complexity maximum-likelihood detection stage to recover
jointly the co-spread users’ data using minimum Euclidean distance
measure and users’ channel gain coefﬁcients. In CDM signal security
is high.
HAND OFF
In cellular telecommunications, the term handover or handoff refers
to the process of transferring an ongoing call or data session from one
channel connected to the core network to another channel. In satellite
communications it is the process of transferring satellite control
responsibility from one earth station to another without loss or
interruption of service.
Purpose
In telecommunications there may be different reasons why a handover

might be conducted:
 when the phone is moving away from the area covered by one
cell and entering the area covered by another cell the call is
transferred to the second cell in order to avoid call termination
when the phone gets outside the range of the first cell;
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 when the capacity for connecting new calls of a given cell is

used up and an existing or new call from a phone, which is
located in an area overlapped by another cell, is transferred to
that cell in order to free-up some capacity in the first cell for
other users, who can only be connected to that cell;
 in non-CDMA networks when the channel used by the phone
becomes interfered by another phone using the same channel in
a different cell, the call is transferred to a different channel in
the same cell or to a different channel in another cell in order to
avoid the interference;
 again in non-CDMA networks when the user behaviour
changes, e.g. when a fast-travelling user, connected to a large,
umbrella-type of cell, stops then the call may be transferred to a
smaller macro cell or even to a micro cell in order to free
capacity on the umbrella cell for other fast-traveling users and to
reduce the potential interference to other cells or users (this
works in reverse too, when a user is detected to be moving faster
than a certain threshold, the call can be transferred to a larger
umbrella-type of cell in order to minimize the frequency of the
handovers due to this movement);
 in CDMA networks a handover (see further down) may be
induced in order to reduce the interference to a smaller
neighboring cell due to the "near-far" effect even when the
phone still has an excellent connection to its current cell;
 etc.
The most basic form of handover is when a phone call in progress is

redirected from its current cell (called source) to a new cell (called
target). In terrestrial networks the source and the target cells may be
served from two different cell sites or from one and the same cell site
(in the latter case the two cells are usually referred to as two sectors
on that cell site). Such a handover, in which the source and the target
are different cells (even if they are on the same cell site) is called
inter-cell handover. The purpose of inter-cell handover is to maintain
the call as the subscriber is moving out of the area covered by the
source cell and entering the area of the target cell.
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A special case is possible, in which the source and the target are one
and the same cell and only the used channel is changed during the
handover. Such a handover, in which the cell is not changed, is called
intra-cell handover. The purpose of intra-cell handover is to change
one channel, which may be interfered or fading with a new clearer or
less fading channel.
Types of handover
In addition to the above classification of inter-cell and intra-cell

classification of handovers, they also can be divided into hard and soft
handovers:
 A hard handover is one in which the channel in the source cell is

released and only then the channel in the target cell is engaged.
Thus the connection to the source is broken before or 'as' the
connection to the target is made—for this reason such handovers
are also known as break-before-make. Hard handovers are
intended to be instantaneous in order to minimize the disruption
to the call. A hard handover is perceived by network engineers
as an event during the call. It requires the least processing by the
network providing service. When the mobile is between base
stations, then the mobile can switch with any of the base
stations, so the base stations bounce the link with the mobile
back and forth. This is called ping-ponging.
 A soft handover is one in which the channel in the source cell is
retained and used for a while in parallel with the channel in the
target cell. In this case the connection to the target is established
before the connection to the source is broken, hence this
handover is called make-before-break. The interval, during
which the two connections are used in parallel, may be brief or
substantial. For this reason the soft handover is perceived by
network engineers as a state of the call, rather than a brief event.
Soft handovers may involve using connections to more than two
cells: connections to three, four or more cells can be maintained
by one phone at the same time. When a call is in a state of soft
handover, the signal of the best of all used channels can be used
for the call at a given moment or all the signals can be combined
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to produce a clearer copy of the signal. The latter is more

advantageous, and when such combining is performed both in
the downlink (forward link) and the uplink (reverse link) the
handover is termed as softer. Softer handovers are possible
when the cells involved in the handovers have a single cell site.
Comparison of handovers
An advantage of the hard handover is that at any moment in time one

call uses only one channel. The hard handover event is indeed very
short and usually is not perceptible by the user. In the old analog
systems it could be heard as a click or a very short beep; in digital
systems it is unnoticeable. Another advantage of the hard handoff is
that the phone's hardware does not need to be capable of receiving
two or more channels in parallel, which makes it cheaper and simpler.
A disadvantage is that if a handover fails the call may be temporarily
disrupted or even terminated abnormally. Technologies which use
hard handovers, usually have procedures which can re-establish the
connection to the source cell if the connection to the target cell cannot
be made. However re-establishing this connection may not always be
possible (in which case the call will be terminated) and even when
possible the procedure may cause a temporary interruption to the call.
One advantage of the soft handovers is that the connection to the

source cell is broken only when a reliable connection to the target cell
has been established and therefore the chances that the call will be
terminated abnormally due to failed handovers are lower. However,
by far a bigger advantage comes from the mere fact that
simultaneously channels in multiple cells are maintained and the call
could only fail if all of the channels are interfered or fade at the same
time. Fading and interference in different channels are unrelated and
therefore the probability of them taking place at the same moment in
all channels is very low. Thus the reliability of the connection
becomes higher when the call is in a soft handover. Because in a
cellular network the majority of the handovers occur in places of poor
coverage, where calls would frequently become unreliable when their
channel is interfered or fading, soft handovers bring a significant
improvement to the reliability of the calls in these places by making
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the interference or the fading in a single channel not critical. This

advantage comes at the cost of more complex hardware in the phone,
which must be capable of processing several channels in parallel.
Another price to pay for soft handovers is use of several channels in
the network to support just a single call. This reduces the number of
remaining free channels and thus reduces the capacity of the network.
By adjusting the duration of soft handovers and the size of the areas in
which they occur, the network engineers can balance the benefit of
extra call reliability against the price of reduced capacity.
Possibility of handover
While theoretically speaking soft handovers are possible in any

technology, analog or digital, the cost of implementing them for
analog technologies is prohibitively high and none of the technologies
that were commercially successful in the past (e.g. AMPS, TACS,
NMT, etc.) had this feature. Of the digital technologies, those based
on FDMA also face a higher cost for the phones (due to the need to
have multiple parallel radio-frequency modules) and those based on
TDMA or a combination of TDMA/FDMA, in principle, allow not so
expensive implementation of soft handovers. However, none of the
2G (second-generation) technologies have this feature (e.g. GSM, D-
AMPS/IS-136, etc.). On the other hand, all CDMA based
technologies, 2G and 3G (third-generation), have soft handovers. On
one hand, this is facilitated by the possibility to design not so
expensive phone hardware supporting soft handovers for CDMA and
on the other hand, this is necessitated by the fact that without soft
handovers CDMA networks may suffer from substantial interference
arising due to the so-called near-far effect.
In all current commercial technologies based on FDMA or on a

combination of TDMA/FDMA (e.g. GSM, AMPS, IS-136/DAMPS,
etc.) changing the channel during a hard handover is realised by
changing the pair of used transmit/receive frequencies.
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Implementations
For the practical realisation of handoffs in a cellular network each cell

is assigned a list of potential target cells, which can be used for
handing-off calls from this source cell to them. These potential target
cells are called neighbours and the list is called neighbour list.
Creating such a list for a given cell is not trivial and specialised
computer tools are used. They implement different algorithms and
may use for input data from field measurements or computer
predictions of radio wave propagation in the areas covered by the
cells.
During a call one or more parameters of the signal in the channel in

the source cell are monitored and assessed in order to decide when a
handover may be necessary. The downlink (forward link) and/or
uplink (reverse link) directions may be monitored. The handover may
be requested by the phone or by the base station (BTS) of its source
cell and, in some systems, by a BTS of a neighbouring cell. The
phone and the BTSs of the neighbouring cells monitor each other
others' signals and the best target candidates are selected among the
neighbouring cells. In some systems, mainly based on CDMA, a
target candidate may be selected among the cells which are not in the
neighbour list. This is done in an effort to reduce the probability of
interference due to the aforementioned near-far effect.
In analog systems the parameters used as criteria for requesting a hard

handover are usually the received signal power and the received
signal-to-noise ratio (the latter may be estimated in an analog system
by inserting additional tones, with frequencies just outside the
captured voice-frequency band at the transmitter and assessing the
form of these tones at the receiver). In non-CDMA 2G digital systems
the criteria for requesting hard handover may be based on estimates of
the received signal power, bit error rate (BER) and block error/erasure
rate (BLER), received quality of speech (RxQual), distance between
the phone and the BTS (estimated from the radio signal propagation
delay) and others. In CDMA systems, 2G and 3G, the most common
criterion for requesting a handover is Ec/Io ratio measured in the pilot
channel (CPICH) and/or RSCP.
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In CDMA systems, when the phone in soft or softer handoff is

connected to several cells simultaneously, it processes the received in
parallel signals using a rake receiver. Each signal is processed by a
module called rake finger. A usual design of a rake receiver in mobile
phones includes three or more rake fingers used in soft handoff state
for processing signals from as many cells and one additional finger
used to search for signals from other cells. The set of cells, whose
signals are used during a soft handoff, is referred to as the active set.
If the search finger finds a sufficiently-strong signal (in terms of high
Ec/Io or RSCP) from a new cell this cell is added to the active set.
The cells in the neighbour list (called in CDMA neighbouring set) are
checked more frequently than the rest and thus a handoff with a
neighbouring cell is more likely, however a handoff with others cells
outside the neighbor list is also allowed (unlike in GSM, IS-
136/DAMPS, AMPS, NMT, etc.).
Satellite
From Wikipedia, the free encyclopedia

This article is about artificial satellites. For natural satellites, also
known as moons, see Natural satellite. For other uses, see Satellite
(disambiguation).
This article is outdated. Please update this article to reflect

recent events or newly available information. (December 2013)
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Play media
NASA's Earth-observing fleet as of June 2012.
An animation depicting the orbits of GPS satellites in medium Earth

orbit.
A full-size model of the Earth observation satellite ERS 2
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In the context of spaceflight, a satellite is an artificial object which

has been intentionally placed into orbit. Such objects are sometimes
called artificial satellites to distinguish them from natural satellites
such as the Moon.
The world's first artificial satellite, the Sputnik 1, was launched by the
Soviet Union in 1957. Since then, thousands of satellites have been
launched into orbit around the Earth. Some satellites, notably space
stations, have been launched in parts and assembled in orbit. Artificial
satellites originate from more than 50 countries and have used the
satellite launching capabilities of ten nations. A few hundred satellites
are currently operational, whereas thousands of unused satellites and
satellite fragments orbit the Earth as space debris. A few space probes
have been placed into orbit around other bodies and become artificial
satellites to the Moon, Mercury, Venus, Mars, Jupiter, Saturn, Vesta,
Eros, and the Sun.
Satellites are used for a large number of purposes. Common types

include military and civilian Earth observation satellites,
communications satellites, navigation satellites, weather satellites, and
research satellites. Space stations and human spacecraft in orbit are
also satellites. Satellite orbits vary greatly, depending on the purpose
of the satellite, and are classified in a number of ways. Well-known
(overlapping) classes include low Earth orbit, polar orbit, and
geostationary orbit.
About 6,600 satellites have been launched. The latest estimates are
that 3,600 remain in orbit.[1] Of those, about 1,000 are operational;[2][3]
the rest have lived out their useful lives and are part of the space
debris. Approximately 500 operational satellites are in low-Earth
orbit, 50 are in medium-Earth orbit (at 20,000 km), the rest are in
geostationary orbit (at 36,000 km).[4]
Satellites are propelled by rockets to their orbits. Usually the launch

vehicle itself is a rocket lifting off from a launch pad on land. In a
minority of cases satellites are launched at sea (from a submarine or a
mobile maritime platform) or aboard a plane (see air launch to orbit).
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Satellites are usually semi-independent computer-controlled systems.

Satellite subsystems attend many tasks, such as power generation,
thermal control, telemetry, attitude control and orbit control.
History of artificial satellites
Sputnik 1: The first artificial satellite to orbit Earth.
The first artificial satellite was Sputnik 1, launched by the Soviet

Union on October 4, 1957, and initiating the Soviet Sputnik program,
with Sergei Korolev as chief designer (there is a crater on the lunar far
side which bears his name). This in turn triggered the Space Race
between the Soviet Union and the United States.
Sputnik 1 helped to identify the density of high atmospheric layers

through measurement of its orbital change and provided data on
radio-signal distribution in the ionosphere. The unanticipated
announcement of Sputnik 1's success precipitated the Sputnik crisis in
the United States and ignited the so-called Space Race within the
Cold War.
Sputnik 2 was launched on November 3, 1957 and carried the first

living passenger into orbit, a dog named Laika.[9]
In May, 1946, Project RAND had released the Preliminary Design of

an Experimental World-Circling Spaceship, which stated, "A satellite
vehicle with appropriate instrumentation can be expected to be one of
the most potent scientific tools of the Twentieth Century."[10] The
United States had been considering launching orbital satellites since
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1945 under the Bureau of Aeronautics of the United States Navy. The
United States Air Force's Project RAND eventually released the
above report, but did not believe that the satellite was a potential
military weapon; rather, they considered it to be a tool for science,
politics, and propaganda. In 1954, the Secretary of Defense stated, "I
know of no American satellite program."[11]
On July 29, 1955, the White House announced that the U.S. intended
to launch satellites by the spring of 1958. This became known as
Project Vanguard. On July 31, the Soviets announced that they
intended to launch a satellite by the fall of 1957.
Following pressure by the American Rocket Society, the National

Science Foundation, and the International Geophysical Year, military
interest picked up and in early 1955 the Army and Navy were
working on Project Orbiter, two competing programs: the army's
which involved using a Jupiter C rocket, and the civilian/Navy
Vanguard Rocket, to launch a satellite. At first, they failed: initial
preference was given to the Vanguard program, whose first attempt at
orbiting a satellite resulted in the explosion of the launch vehicle on
national television. But finally, three months after Sputnik 2, the
project succeeded; Explorer 1 became the United States' first artificial
satellite on January 31, 1958.[12]
In June 1961, three-and-a-half years after the launch of Sputnik 1, the

Air Force used resources of the United States Space Surveillance
Network to catalog 115 Earth-orbiting satellites.[13]
Early satellites were constructed as "one-off" designs. With growth in

geosynchronous (GEO) satellite communication, multiple satellites
began to be built on single model platforms called satellite buses. The
first standardized satellite bus design was the HS-333 GEO commsat,
launched in 1972.
The largest artificial satellite currently orbiting the Earth is the

International Space Station.
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1U CubeSat ESTCube-1, developed mainly by the students from the

University of Tartu, carries out a tether deployment experiment on the
low Earth orbit.
Space Surveillance Network
Main article: United States Space Surveillance Network
The United States Space Surveillance Network (SSN), a division of

The United States Strategic Command, has been tracking objects in
Earth's orbit since 1957 when the Soviets opened the space age with
the launch of Sputnik I. Since then, the SSN has tracked more than
26,000 objects. The SSN currently tracks more than 8,000 man-made
orbiting objects. The rest have re-entered Earth's atmosphere and
disintegrated, or survived re-entry and impacted the Earth. The SSN
tracks objects that are 10 centimeters in diameter or larger; those now
orbiting Earth range from satellites weighing several tons to pieces of
spent rocket bodies weighing only 10 pounds. About seven percent
are operational satellites (i.e. ~560 satellites), the rest are space
debris.[14] The United States Strategic Command is primarily
interested in the active satellites, but also tracks space debris which
upon reentry might otherwise be mistaken for incoming missiles.
A search of the NSSDC Master Catalog at the end of October 2010

listed 6,578 satellites launched into orbit since 1957, the latest being
Chang'e 2, on 1 October 2010.[15]
Non-military satellite services
There are three basic categories of non-military satellite services:[16]
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Fixed satellite services
Fixed satellite services handle hundreds of billions of voice, data, and

video transmission tasks across all countries and continents between
certain points on the Earth's surface.
Mobile satellite systems
Mobile satellite systems help connect remote regions, vehicles, ships,

people and aircraft to other parts of the world and/or other mobile or
stationary communications units, in addition to serving as navigation
systems.
Scientific research satellites (commercial and noncommercial)
Scientific research satellites provide meteorological information, land

survey data (e.g. remote sensing), Amateur (HAM) Radio, and other
different scientific research applications such as earth science, marine
science, and atmospheric research.
Types
MILSTAR: A communication satellite
 "Killer Satellites" are satellites that are designed to destroy

enemy warheads, satellites, and other space assets.
 Astronomical satellites are satellites used for observation of
distant planets, galaxies, and other outer space objects.
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 Biosatellites are satellites designed to carry living organisms,

generally for scientific experimentation.
 Communications satellites are satellites stationed in space for
the purpose of telecommunications. Modern communications
satellites typically use geosynchronous orbits, Molniya orbits or
Low Earth orbits.
 Miniaturized satellites are satellites of unusually low masses
and small sizes.[17] New classifications are used to categorize
these satellites: minisatellite (500–100 kg), microsatellite (below
100 kg), nanosatellite (below 10 kg).[citation needed]
 Navigational satellites are satellites which use radio time
signals transmitted to enable mobile receivers on the ground to
determine their exact location. The relatively clear line of sight
between the satellites and receivers on the ground, combined
with ever-improving electronics, allows satellite navigation
systems to measure location to accuracies on the order of a few
meters in real time.
 Reconnaissance satellites are Earth observation satellite or
communications satellite deployed for military or intelligence
applications. Very little is known about the full power of these
satellites, as governments who operate them usually keep
information pertaining to their reconnaissance satellites
classified.
 Earth observation satellites are satellites intended for non-
military uses such as environmental monitoring, meteorology,
map making etc. (See especially Earth Observing System.)
 Tether satellites are satellites which are connected to another
satellite by a thin cable called a tether.
 Weather satellites are primarily used to monitor Earth's
weather and climate.[18]
 Recovery satellites are satellites that provide a recovery of
reconnaissance, biological, space-production and other payloads
from orbit to Earth.
 Manned spacecraft (spaceships) are large satellites able to put
humans into (and beyond) an orbit, and return them to Earth.
Spacecraft including spaceplanes of reusable systems have
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major propulsion or landing facilities. They can be used as

transport to and from the orbital stations.
International Space Station as seen from Space
 Space stations are man-made orbital structures that are

designed for human beings to live on in outer space. A space
station is distinguished from other manned spacecraft by its lack
of major propulsion or landing facilities. Space stations are
designed for medium-term living in orbit, for periods of weeks,
months, or even years.
 A Skyhook is a proposed type of tethered satellite/ion powered
space station that serves as a terminal for suborbital launch
vehicles flying between the Earth and the lower end of the
Skyhook, as well as a terminal for spacecraft going to, or
arriving from, higher orbit, the Moon, or Mars, at the upper end
of the Skyhook.[19][20]
BLUETOOTH
This article is about a wireless technology standard. For the medieval
King of Denmark, see Harald Bluetooth.
Bluetooth
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Developed Bluetooth Special Interest

by Group
Mobile personal area

Industry
networks
Mobile phones, Personal

Compatible
computers, Laptop
hardware
computers
Physical
Up to 60 metres[1]
range
Bluetooth is a wireless technology standard for exchanging data over

short distances (using short-wavelength UHF radio waves in the ISM
band from 2.4 to 2.485 GHz[2]) from fixed and mobile devices, and
building personal area networks (PANs). Invented by telecom vendor
Ericsson in 1994,[3] it was originally conceived as a wireless
alternative to RS-232 data cables. It can connect several devices,
overcoming problems of synchronization.
Bluetooth is managed by the Bluetooth Special Interest Group (SIG),

which has more than 20,000 member companies in the areas of
telecommunication, computing, networking, and consumer
electronics.[4] Bluetooth was standardized as IEEE 802.15.1, but the
standard is no longer maintained. The SIG oversees the development
of the specification, manages the qualification program, and protects
the trademarks.[5] To be marketed as a Bluetooth device, it must be
qualified to standards defined by the SIG.[6] A network of patents is
required to implement the technology, which is licensed only for that
qualifying device.
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Implementation
Bluetooth operates in the range of 2400–2483.5 MHz (including

guard bands). This is in the globally unlicensed (but not unregulated)
Industrial, Scientific and Medical (ISM) 2.4 GHz short-range radio
frequency band. Bluetooth uses a radio technology called frequency-
hopping spread spectrum. The transmitted data are divided into
packets and each packet is transmitted on one of the 79 designated
Bluetooth channels. Each channel has a bandwidth of 1 MHz.
Bluetooth 4.0 uses 2 MHz spacing which allows for 40 channels. The
first channel starts at 2402 MHz and continues up to 2480 MHz in
1 MHz steps. It usually performs 1600 hops per second, with
Adaptive Frequency-Hopping (AFH) enabled.[12]
Originally, Gaussian frequency-shift keying (GFSK) modulation was

the only modulation scheme available; subsequently, since the
introduction of Bluetooth 2.0+EDR, π/4-DQPSK and 8DPSK
modulation may also be used between compatible devices. Devices
functioning with GFSK are said to be operating in basic rate (BR)
mode where an instantaneous data rate of 1 Mbit/s is possible. The
term Enhanced Data Rate (EDR) is used to describe π/4-DPSK and
8DPSK schemes, each giving 2 and 3 Mbit/s respectively. The
combination of these (BR and EDR) modes in Bluetooth radio
technology is classified as a "BR/EDR radio".
Bluetooth is a packet-based protocol with a master-slave structure.

One master may communicate with up to seven slaves in a piconet; all
devices share the master's clock. Packet exchange is based on the
basic clock, defined by the master, which ticks at 312.5 µs intervals.
Two clock ticks make up a slot of 625 µs; two slots make up a slot
pair of 1250 µs. In the simple case of single-slot packets the master
transmits in even slots and receives in odd slots; the slave, conversely,
receives in even slots and transmits in odd slots. Packets may be 1, 3
or 5 slots long, but in all cases the master transmit will begin in even
slots and the slave transmit in odd slots.
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Communication and connection
A master Bluetooth device can communicate with a maximum of

seven devices in a piconet (an ad-hoc computer network using
Bluetooth technology), though not all devices reach this maximum.
The devices can switch roles, by agreement, and the slave can become
the master (for example, a headset initiating a connection to a phone
will necessarily begin as master, as initiator of the connection; but
may subsequently prefer to be slave).
The Bluetooth Core Specification provides for the connection of two

or more piconets to form a scatternet, in which certain devices
simultaneously play the master role in one piconet and the slave role
in another.
At any given time, data can be transferred between the master and one
other device (except for the little-used broadcast mode.[citation needed])
The master chooses which slave device to address; typically, it
switches rapidly from one device to another in a round-robin fashion.
Since it is the master that chooses which slave to address, whereas a
slave is (in theory) supposed to listen in each receive slot, being a
master is a lighter burden than being a slave. Being a master of seven
slaves is possible; being a slave of more than one master is
difficult.[citation needed] The specification is vague as to required behavior
in scatternets.
Many USB Bluetooth adapters or "dongles" are available, some of

which also include an IrDA adapter.[citation needed]
Uses
Max. permitted power Typ. range[13]
Class
(mW) (dBm) (m)
1 100 20 ~100
2 2.5 4 ~10
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3 1 0 ~1
Bluetooth is a standard wire-replacement communications protocol

primarily designed for low-power consumption, with a short range
based on low-cost transceiver microchips in each device.[14] Because
the devices use a radio (broadcast) communications system, they do
not have to be in visual line of sight of each other, however a quasi
optical wireless path must be viable.[4] Range is power-class-
dependent, but effective ranges vary in practice; see the table on the
right.
Version Data rate Max. application throughput
1.2 1 Mbit/s >80 kbit/s
2.0 + EDR 3 Mbit/s >80 kbit/s
3.0 + HS 24 Mbit/s See Version 3.0 + HS
4.0 24 Mbit/s See Version 4.0 LE
The effective range varies due to propagation conditions, material

coverage, production sample variations, antenna configurations and
battery conditions. Most Bluetooth applications are in indoor
conditions, where attenuation of walls and signal fading due to signal
reflections will cause the range to be far lower than the specified line-
of-sight ranges of the Bluetooth products. Most Bluetooth
applications are battery powered Class 2 devices, with little difference
in range whether the other end of the link is a Class 1 or Class 2
device as the lower powered device tends to set the range limit. In
some cases the effective range of the data link can be extended when
a Class 2 devices is connecting to a Class 1 transceiver with both
higher sensitivity and transmission power than a typical Class 2
device.[15] Mostly however the Class 1 devices have a similar
sensitivity to Class 2 devices. Connecting two Class 1 devices with
both high sensitivity and high power can allow ranges far in excess of
the typical 100m, depending on the throughput required by the
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application. Some such devices allow open field ranges of up to 1 km

and beyond between two similar devices without exceeding legal
emission limits.[16][17][18]
While the Bluetooth Core Specification does mandate minimal for

range, the range of the technology is application-specific and not
limited. Manufacturers may tune their implementations to the range
needed for individual use cases.
Bluetooth profiles
Main article: Bluetooth profile
To use Bluetooth wireless technology, a device has to be able to

interpret certain Bluetooth profiles, which are definitions of possible
applications and specify general behaviours that Bluetooth enabled
devices use to communicate with other Bluetooth devices. These
profiles include settings to parametrize and to control the
communication from start. Adherence to profiles saves the time for
transmitting the parameters anew before the bi-directional link
becomes effective. There are a wide range of Bluetooth profiles that
describe many different types of applications or use cases for
devices.[19][20]
List of applications
A typical Bluetooth mobile phone headset.
 Wireless control of and communication between a mobile phone

and a handsfree headset. This was one of the earliest
applications to become popular.
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 Wireless control of and communication between a mobile phone

and a Bluetooth compatible car stereo system.
 Wireless control of and communication with tablets and
speakers such as iPad and Android devices.
 Wireless Bluetooth headset and Intercom. Idiomatically, a
headset is sometimes called "a Bluetooth".
 Wireless networking between PCs in a confined space and
where little bandwidth is required.
 Wireless communication with PC input and output devices, the
most common being the mouse, keyboard and printer.
 Transfer of files, contact details, calendar appointments, and
reminders between devices with OBEX.
 Replacement of previous wired RS-232 serial communications
in test equipment, GPS receivers, medical equipment, bar code
scanners, and traffic control devices.
 For controls where infrared was often used.
 For low bandwidth applications where higher USB bandwidth is
not required and cable-free connection desired.
 Sending small advertisements from Bluetooth-enabled
advertising hoardings to other, discoverable, Bluetooth
devices.[21]
 Wireless bridge between two Industrial Ethernet (e.g.,
PROFINET) networks.
 Three seventh and eighth generation game consoles, Nintendo's
Wii.[22] and Sony's PlayStation 3, use Bluetooth for their
respective wireless controllers.
 Dial-up internet access on personal computers or PDAs using a
data-capable mobile phone as a wireless modem.
 Short range transmission of health sensor data from medical
devices to mobile phone, set-top box or dedicated telehealth
devices.[23]
 Allowing a DECT phone to ring and answer calls on behalf of a
nearby mobile phone.
 Real-time location systems (RTLS), are used to track and
identify the location of objects in real-time using “Nodes” or
“tags” attached to, or embedded in the objects tracked, and
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“Readers” that receive and process the wireless signals from

these tags to determine their locations.[24]
 Personal security application on mobile phones for prevention of
theft or loss of items. The protected item has a Bluetooth marker
(e.g., a tag) that is in constant communication with the phone. If
the connection is broken (the marker is out of range of the
phone) then an alarm is raised. This can also be used as a man
overboard alarm. A product using this technology has been
available since 2009.[25]
 Calgary, Alberta, Canada's Roads Traffic division uses data
collected from travelers' Bluetooth devices to predict travel
times and road congestion for motorists.[26]
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UNIT I FUNDAMENTALS OF ANALOG COMMUNICATION

UNIT II DIGITAL COMMUNICATION

FREQUENCY SHIFT KEYING:

UNIT III DIGITAL TRANSMISSION

Delta Modulation (DM) :

Let m[ n ] = m(nT s ) , n = 0,±1,±2, 

The modulator consists of a comparator, a quantizer, and an accumulator. The output of

Differential Pulse-Code Modulation (DPCM):

Figure 3.28 DPCM system. (a) Transmitter. (b) Receiver.

The (SNR) o of the DPCM system is

where σ M2 and σQ2 are variances of m[ n ] ( E[ m[ n]] = 0) and q[ n ]

NEAR EAST UNIVERSITY

CH. 1: INTRODUCTION TO DATA COMMUNICATIONS

Basic Elements of a Communication System

Digital data takes on discrete values; eg. a computer’s output.

 Two ways of transmitting analog information. In either cases it is not data

Data Communication System Channel

Output Output Source Channel Digital

Figure 1.2. Basic elements of a digital communication system

Early Work in Digital Communications

 Frequency, Spectrum, & Bandwidth

 Analog and Digital Signals

Analog Signal Digital Signal

 Time –Domain Concepts

lim sc (t )  s(t ) for all a.

s(t )  sin 2ft  1 / 3 sin 3(2ft )  1 / 5 sin 5(2ft )

1/3 sin 3(2ft)

1/5 sin 5(2ft)

and an infinite bandwidth as shown below:

1/T 2/T 3/T 4/T n/T

With a dc component, it has a frequency term at f = 0 and a non-zero average amplitude.

s1 (t )  1.2  sin (2ft )  1 / 3 sin 3(2ft )  1 / 5 sin 5(2ft )

Relationship between Data Rate and Bandwidth

sin (2ft )  1 / 3 sin(2 (3 f )t )  1 / 5 sin(2 (5 f )t )

sin (2ft )  1/ 3sin(2 (3 f )t )

Amplitude, Period and Phase

Bit Interval and Bit Rate

Decomposition of a Digital Signal

a) First harmonic only b) First, third, and fifth harmonics

c) First, third, fifth, and seventh harmonics d) Infinite number of harmonics

a) Spectrum for exact replica

Transmission Media: Guided

There four basic types of guided media:

Unshielded Twisted Pair

The degree of reduction in noise interference is determined specifically by the number of

Shielded Twisted Pair

Fibre Optic Cables

All fibres consist of a number of substructures including:

A core, which carries most of the light, surrounded by

where n1-refractive index of material 1; 1-angle of incidence; 2-angle of refraction;

internal reflection occurs as it strikes the lower index cladding material.

Basic Construction of Fibre-Optic Cables

1. Multimode Step-Index Fibre.

Cross section Index profile Beam path dispersion

2. Single Mode Cable

Cross section Index profile Beam path

3. Multimode graded-index fibre cables

Cross section Index profile Beam path